AudioFlinger.cpp revision 3b86c964df855a9740c446e984309b719c3ec37c
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 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 39#include <media/AudioTrack.h> 40#include <media/AudioRecord.h> 41#include <media/IMediaPlayerService.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <cpustats/ThreadCpuUsage.h> 58#include <powermanager/PowerManager.h> 59// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 60 61// ---------------------------------------------------------------------------- 62 63 64namespace android { 65 66static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 67static const char* kHardwareLockedString = "Hardware lock is taken\n"; 68 69//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 70static const float MAX_GAIN = 4096.0f; 71static const float MAX_GAIN_INT = 0x1000; 72 73// retry counts for buffer fill timeout 74// 50 * ~20msecs = 1 second 75static const int8_t kMaxTrackRetries = 50; 76static const int8_t kMaxTrackStartupRetries = 50; 77// allow less retry attempts on direct output thread. 78// direct outputs can be a scarce resource in audio hardware and should 79// be released as quickly as possible. 80static const int8_t kMaxTrackRetriesDirect = 2; 81 82static const int kDumpLockRetries = 50; 83static const int kDumpLockSleep = 20000; 84 85static const nsecs_t kWarningThrottle = seconds(5); 86 87// RecordThread loop sleep time upon application overrun or audio HAL read error 88static const int kRecordThreadSleepUs = 5000; 89 90static const nsecs_t kSetParametersTimeout = seconds(2); 91 92// ---------------------------------------------------------------------------- 93 94static bool recordingAllowed() { 95 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 96 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 97 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 98 return ok; 99} 100 101static bool settingsAllowed() { 102 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 103 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 104 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 105 return ok; 106} 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IBinder> binder = 111 defaultServiceManager()->getService(String16("media.player")); 112 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 113 if (service.get() == NULL) { 114 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 115 return; 116 } 117 118 service->addBatteryData(params); 119} 120 121static int load_audio_interface(const char *if_name, const hw_module_t **mod, 122 audio_hw_device_t **dev) 123{ 124 int rc; 125 126 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 127 if (rc) 128 goto out; 129 130 rc = audio_hw_device_open(*mod, dev); 131 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 132 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 133 if (rc) 134 goto out; 135 136 return 0; 137 138out: 139 *mod = NULL; 140 *dev = NULL; 141 return rc; 142} 143 144static const char *audio_interfaces[] = { 145 "primary", 146 "a2dp", 147 "usb", 148}; 149#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 150 151// ---------------------------------------------------------------------------- 152 153AudioFlinger::AudioFlinger() 154 : BnAudioFlinger(), 155 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 156 mBtNrecIsOff(false) 157{ 158} 159 160void AudioFlinger::onFirstRef() 161{ 162 int rc = 0; 163 164 Mutex::Autolock _l(mLock); 165 166 /* TODO: move all this work into an Init() function */ 167 mHardwareStatus = AUDIO_HW_IDLE; 168 169 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 170 const hw_module_t *mod; 171 audio_hw_device_t *dev; 172 173 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 174 if (rc) 175 continue; 176 177 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 178 mod->name, mod->id); 179 mAudioHwDevs.push(dev); 180 181 if (!mPrimaryHardwareDev) { 182 mPrimaryHardwareDev = dev; 183 LOGI("Using '%s' (%s.%s) as the primary audio interface", 184 mod->name, mod->id, audio_interfaces[i]); 185 } 186 } 187 188 mHardwareStatus = AUDIO_HW_INIT; 189 190 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 191 LOGE("Primary audio interface not found"); 192 return; 193 } 194 195 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 196 audio_hw_device_t *dev = mAudioHwDevs[i]; 197 198 mHardwareStatus = AUDIO_HW_INIT; 199 rc = dev->init_check(dev); 200 if (rc == 0) { 201 AutoMutex lock(mHardwareLock); 202 203 mMode = AUDIO_MODE_NORMAL; 204 mHardwareStatus = AUDIO_HW_SET_MODE; 205 dev->set_mode(dev, mMode); 206 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 207 dev->set_master_volume(dev, 1.0f); 208 mHardwareStatus = AUDIO_HW_IDLE; 209 } 210 } 211} 212 213status_t AudioFlinger::initCheck() const 214{ 215 Mutex::Autolock _l(mLock); 216 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 217 return NO_INIT; 218 return NO_ERROR; 219} 220 221AudioFlinger::~AudioFlinger() 222{ 223 int num_devs = mAudioHwDevs.size(); 224 225 while (!mRecordThreads.isEmpty()) { 226 // closeInput() will remove first entry from mRecordThreads 227 closeInput(mRecordThreads.keyAt(0)); 228 } 229 while (!mPlaybackThreads.isEmpty()) { 230 // closeOutput() will remove first entry from mPlaybackThreads 231 closeOutput(mPlaybackThreads.keyAt(0)); 232 } 233 234 for (int i = 0; i < num_devs; i++) { 235 audio_hw_device_t *dev = mAudioHwDevs[i]; 236 audio_hw_device_close(dev); 237 } 238 mAudioHwDevs.clear(); 239} 240 241audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 242{ 243 /* first matching HW device is returned */ 244 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 if ((dev->get_supported_devices(dev) & devices) == devices) 247 return dev; 248 } 249 return NULL; 250} 251 252status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 253{ 254 const size_t SIZE = 256; 255 char buffer[SIZE]; 256 String8 result; 257 258 result.append("Clients:\n"); 259 for (size_t i = 0; i < mClients.size(); ++i) { 260 wp<Client> wClient = mClients.valueAt(i); 261 if (wClient != 0) { 262 sp<Client> client = wClient.promote(); 263 if (client != 0) { 264 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 265 result.append(buffer); 266 } 267 } 268 } 269 270 result.append("Global session refs:\n"); 271 result.append(" session pid cnt\n"); 272 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 273 AudioSessionRef *r = mAudioSessionRefs[i]; 274 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 275 result.append(buffer); 276 } 277 write(fd, result.string(), result.size()); 278 return NO_ERROR; 279} 280 281 282status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 283{ 284 const size_t SIZE = 256; 285 char buffer[SIZE]; 286 String8 result; 287 int hardwareStatus = mHardwareStatus; 288 289 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 290 result.append(buffer); 291 write(fd, result.string(), result.size()); 292 return NO_ERROR; 293} 294 295status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 296{ 297 const size_t SIZE = 256; 298 char buffer[SIZE]; 299 String8 result; 300 snprintf(buffer, SIZE, "Permission Denial: " 301 "can't dump AudioFlinger from pid=%d, uid=%d\n", 302 IPCThreadState::self()->getCallingPid(), 303 IPCThreadState::self()->getCallingUid()); 304 result.append(buffer); 305 write(fd, result.string(), result.size()); 306 return NO_ERROR; 307} 308 309static bool tryLock(Mutex& mutex) 310{ 311 bool locked = false; 312 for (int i = 0; i < kDumpLockRetries; ++i) { 313 if (mutex.tryLock() == NO_ERROR) { 314 locked = true; 315 break; 316 } 317 usleep(kDumpLockSleep); 318 } 319 return locked; 320} 321 322status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 323{ 324 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 325 dumpPermissionDenial(fd, args); 326 } else { 327 // get state of hardware lock 328 bool hardwareLocked = tryLock(mHardwareLock); 329 if (!hardwareLocked) { 330 String8 result(kHardwareLockedString); 331 write(fd, result.string(), result.size()); 332 } else { 333 mHardwareLock.unlock(); 334 } 335 336 bool locked = tryLock(mLock); 337 338 // failed to lock - AudioFlinger is probably deadlocked 339 if (!locked) { 340 String8 result(kDeadlockedString); 341 write(fd, result.string(), result.size()); 342 } 343 344 dumpClients(fd, args); 345 dumpInternals(fd, args); 346 347 // dump playback threads 348 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 349 mPlaybackThreads.valueAt(i)->dump(fd, args); 350 } 351 352 // dump record threads 353 for (size_t i = 0; i < mRecordThreads.size(); i++) { 354 mRecordThreads.valueAt(i)->dump(fd, args); 355 } 356 357 // dump all hardware devs 358 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 359 audio_hw_device_t *dev = mAudioHwDevs[i]; 360 dev->dump(dev, fd); 361 } 362 if (locked) mLock.unlock(); 363 } 364 return NO_ERROR; 365} 366 367 368// IAudioFlinger interface 369 370 371sp<IAudioTrack> AudioFlinger::createTrack( 372 pid_t pid, 373 int streamType, 374 uint32_t sampleRate, 375 uint32_t format, 376 uint32_t channelMask, 377 int frameCount, 378 uint32_t flags, 379 const sp<IMemory>& sharedBuffer, 380 int output, 381 int *sessionId, 382 status_t *status) 383{ 384 sp<PlaybackThread::Track> track; 385 sp<TrackHandle> trackHandle; 386 sp<Client> client; 387 wp<Client> wclient; 388 status_t lStatus; 389 int lSessionId; 390 391 if (streamType >= AUDIO_STREAM_CNT) { 392 LOGE("invalid stream type"); 393 lStatus = BAD_VALUE; 394 goto Exit; 395 } 396 397 { 398 Mutex::Autolock _l(mLock); 399 PlaybackThread *thread = checkPlaybackThread_l(output); 400 PlaybackThread *effectThread = NULL; 401 if (thread == NULL) { 402 LOGE("unknown output thread"); 403 lStatus = BAD_VALUE; 404 goto Exit; 405 } 406 407 wclient = mClients.valueFor(pid); 408 409 if (wclient != NULL) { 410 client = wclient.promote(); 411 } else { 412 client = new Client(this, pid); 413 mClients.add(pid, client); 414 } 415 416 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 417 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 418 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 419 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 420 if (mPlaybackThreads.keyAt(i) != output) { 421 // prevent same audio session on different output threads 422 uint32_t sessions = t->hasAudioSession(*sessionId); 423 if (sessions & PlaybackThread::TRACK_SESSION) { 424 lStatus = BAD_VALUE; 425 goto Exit; 426 } 427 // check if an effect with same session ID is waiting for a track to be created 428 if (sessions & PlaybackThread::EFFECT_SESSION) { 429 effectThread = t.get(); 430 } 431 } 432 } 433 lSessionId = *sessionId; 434 } else { 435 // if no audio session id is provided, create one here 436 lSessionId = nextUniqueId(); 437 if (sessionId != NULL) { 438 *sessionId = lSessionId; 439 } 440 } 441 ALOGV("createTrack() lSessionId: %d", lSessionId); 442 443 track = thread->createTrack_l(client, streamType, sampleRate, format, 444 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 445 446 // move effect chain to this output thread if an effect on same session was waiting 447 // for a track to be created 448 if (lStatus == NO_ERROR && effectThread != NULL) { 449 Mutex::Autolock _dl(thread->mLock); 450 Mutex::Autolock _sl(effectThread->mLock); 451 moveEffectChain_l(lSessionId, effectThread, thread, true); 452 } 453 } 454 if (lStatus == NO_ERROR) { 455 trackHandle = new TrackHandle(track); 456 } else { 457 // remove local strong reference to Client before deleting the Track so that the Client 458 // destructor is called by the TrackBase destructor with mLock held 459 client.clear(); 460 track.clear(); 461 } 462 463Exit: 464 if(status) { 465 *status = lStatus; 466 } 467 return trackHandle; 468} 469 470uint32_t AudioFlinger::sampleRate(int output) const 471{ 472 Mutex::Autolock _l(mLock); 473 PlaybackThread *thread = checkPlaybackThread_l(output); 474 if (thread == NULL) { 475 LOGW("sampleRate() unknown thread %d", output); 476 return 0; 477 } 478 return thread->sampleRate(); 479} 480 481int AudioFlinger::channelCount(int output) const 482{ 483 Mutex::Autolock _l(mLock); 484 PlaybackThread *thread = checkPlaybackThread_l(output); 485 if (thread == NULL) { 486 LOGW("channelCount() unknown thread %d", output); 487 return 0; 488 } 489 return thread->channelCount(); 490} 491 492uint32_t AudioFlinger::format(int output) const 493{ 494 Mutex::Autolock _l(mLock); 495 PlaybackThread *thread = checkPlaybackThread_l(output); 496 if (thread == NULL) { 497 LOGW("format() unknown thread %d", output); 498 return 0; 499 } 500 return thread->format(); 501} 502 503size_t AudioFlinger::frameCount(int output) const 504{ 505 Mutex::Autolock _l(mLock); 506 PlaybackThread *thread = checkPlaybackThread_l(output); 507 if (thread == NULL) { 508 LOGW("frameCount() unknown thread %d", output); 509 return 0; 510 } 511 return thread->frameCount(); 512} 513 514uint32_t AudioFlinger::latency(int output) const 515{ 516 Mutex::Autolock _l(mLock); 517 PlaybackThread *thread = checkPlaybackThread_l(output); 518 if (thread == NULL) { 519 LOGW("latency() unknown thread %d", output); 520 return 0; 521 } 522 return thread->latency(); 523} 524 525status_t AudioFlinger::setMasterVolume(float value) 526{ 527 status_t ret = initCheck(); 528 if (ret != NO_ERROR) { 529 return ret; 530 } 531 532 // check calling permissions 533 if (!settingsAllowed()) { 534 return PERMISSION_DENIED; 535 } 536 537 // when hw supports master volume, don't scale in sw mixer 538 { // scope for the lock 539 AutoMutex lock(mHardwareLock); 540 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 541 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 542 value = 1.0f; 543 } 544 mHardwareStatus = AUDIO_HW_IDLE; 545 } 546 547 Mutex::Autolock _l(mLock); 548 mMasterVolume = value; 549 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 550 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 551 552 return NO_ERROR; 553} 554 555status_t AudioFlinger::setMode(int mode) 556{ 557 status_t ret = initCheck(); 558 if (ret != NO_ERROR) { 559 return ret; 560 } 561 562 // check calling permissions 563 if (!settingsAllowed()) { 564 return PERMISSION_DENIED; 565 } 566 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 567 LOGW("Illegal value: setMode(%d)", mode); 568 return BAD_VALUE; 569 } 570 571 { // scope for the lock 572 AutoMutex lock(mHardwareLock); 573 mHardwareStatus = AUDIO_HW_SET_MODE; 574 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 575 mHardwareStatus = AUDIO_HW_IDLE; 576 } 577 578 if (NO_ERROR == ret) { 579 Mutex::Autolock _l(mLock); 580 mMode = mode; 581 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 582 mPlaybackThreads.valueAt(i)->setMode(mode); 583 } 584 585 return ret; 586} 587 588status_t AudioFlinger::setMicMute(bool state) 589{ 590 status_t ret = initCheck(); 591 if (ret != NO_ERROR) { 592 return ret; 593 } 594 595 // check calling permissions 596 if (!settingsAllowed()) { 597 return PERMISSION_DENIED; 598 } 599 600 AutoMutex lock(mHardwareLock); 601 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 602 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 603 mHardwareStatus = AUDIO_HW_IDLE; 604 return ret; 605} 606 607bool AudioFlinger::getMicMute() const 608{ 609 status_t ret = initCheck(); 610 if (ret != NO_ERROR) { 611 return false; 612 } 613 614 bool state = AUDIO_MODE_INVALID; 615 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 616 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 617 mHardwareStatus = AUDIO_HW_IDLE; 618 return state; 619} 620 621status_t AudioFlinger::setMasterMute(bool muted) 622{ 623 // check calling permissions 624 if (!settingsAllowed()) { 625 return PERMISSION_DENIED; 626 } 627 628 Mutex::Autolock _l(mLock); 629 mMasterMute = muted; 630 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 631 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 632 633 return NO_ERROR; 634} 635 636float AudioFlinger::masterVolume() const 637{ 638 return mMasterVolume; 639} 640 641bool AudioFlinger::masterMute() const 642{ 643 return mMasterMute; 644} 645 646status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 647{ 648 // check calling permissions 649 if (!settingsAllowed()) { 650 return PERMISSION_DENIED; 651 } 652 653 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 654 return BAD_VALUE; 655 } 656 657 AutoMutex lock(mLock); 658 PlaybackThread *thread = NULL; 659 if (output) { 660 thread = checkPlaybackThread_l(output); 661 if (thread == NULL) { 662 return BAD_VALUE; 663 } 664 } 665 666 mStreamTypes[stream].volume = value; 667 668 if (thread == NULL) { 669 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 670 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 671 } 672 } else { 673 thread->setStreamVolume(stream, value); 674 } 675 676 return NO_ERROR; 677} 678 679status_t AudioFlinger::setStreamMute(int stream, bool muted) 680{ 681 // check calling permissions 682 if (!settingsAllowed()) { 683 return PERMISSION_DENIED; 684 } 685 686 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 687 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 688 return BAD_VALUE; 689 } 690 691 AutoMutex lock(mLock); 692 mStreamTypes[stream].mute = muted; 693 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 694 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 695 696 return NO_ERROR; 697} 698 699float AudioFlinger::streamVolume(int stream, int output) const 700{ 701 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 702 return 0.0f; 703 } 704 705 AutoMutex lock(mLock); 706 float volume; 707 if (output) { 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 return 0.0f; 711 } 712 volume = thread->streamVolume(stream); 713 } else { 714 volume = mStreamTypes[stream].volume; 715 } 716 717 return volume; 718} 719 720bool AudioFlinger::streamMute(int stream) const 721{ 722 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 723 return true; 724 } 725 726 return mStreamTypes[stream].mute; 727} 728 729status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 730{ 731 status_t result; 732 733 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 734 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 735 // check calling permissions 736 if (!settingsAllowed()) { 737 return PERMISSION_DENIED; 738 } 739 740 // ioHandle == 0 means the parameters are global to the audio hardware interface 741 if (ioHandle == 0) { 742 AutoMutex lock(mHardwareLock); 743 mHardwareStatus = AUDIO_SET_PARAMETER; 744 status_t final_result = NO_ERROR; 745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 746 audio_hw_device_t *dev = mAudioHwDevs[i]; 747 result = dev->set_parameters(dev, keyValuePairs.string()); 748 final_result = result ?: final_result; 749 } 750 mHardwareStatus = AUDIO_HW_IDLE; 751 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 752 AudioParameter param = AudioParameter(keyValuePairs); 753 String8 value; 754 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 755 Mutex::Autolock _l(mLock); 756 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 757 if (mBtNrecIsOff != btNrecIsOff) { 758 for (size_t i = 0; i < mRecordThreads.size(); i++) { 759 sp<RecordThread> thread = mRecordThreads.valueAt(i); 760 RecordThread::RecordTrack *track = thread->track(); 761 if (track != NULL) { 762 audio_devices_t device = (audio_devices_t)( 763 thread->device() & AUDIO_DEVICE_IN_ALL); 764 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 765 thread->setEffectSuspended(FX_IID_AEC, 766 suspend, 767 track->sessionId()); 768 thread->setEffectSuspended(FX_IID_NS, 769 suspend, 770 track->sessionId()); 771 } 772 } 773 mBtNrecIsOff = btNrecIsOff; 774 } 775 } 776 return final_result; 777 } 778 779 // hold a strong ref on thread in case closeOutput() or closeInput() is called 780 // and the thread is exited once the lock is released 781 sp<ThreadBase> thread; 782 { 783 Mutex::Autolock _l(mLock); 784 thread = checkPlaybackThread_l(ioHandle); 785 if (thread == NULL) { 786 thread = checkRecordThread_l(ioHandle); 787 } else if (thread.get() == primaryPlaybackThread_l()) { 788 // indicate output device change to all input threads for pre processing 789 AudioParameter param = AudioParameter(keyValuePairs); 790 int value; 791 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 792 for (size_t i = 0; i < mRecordThreads.size(); i++) { 793 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 794 } 795 } 796 } 797 } 798 if (thread != NULL) { 799 result = thread->setParameters(keyValuePairs); 800 return result; 801 } 802 return BAD_VALUE; 803} 804 805String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 806{ 807// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 808// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 809 810 if (ioHandle == 0) { 811 String8 out_s8; 812 813 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 814 audio_hw_device_t *dev = mAudioHwDevs[i]; 815 char *s = dev->get_parameters(dev, keys.string()); 816 out_s8 += String8(s); 817 free(s); 818 } 819 return out_s8; 820 } 821 822 Mutex::Autolock _l(mLock); 823 824 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 825 if (playbackThread != NULL) { 826 return playbackThread->getParameters(keys); 827 } 828 RecordThread *recordThread = checkRecordThread_l(ioHandle); 829 if (recordThread != NULL) { 830 return recordThread->getParameters(keys); 831 } 832 return String8(""); 833} 834 835size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 836{ 837 status_t ret = initCheck(); 838 if (ret != NO_ERROR) { 839 return 0; 840 } 841 842 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 843} 844 845unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 846{ 847 if (ioHandle == 0) { 848 return 0; 849 } 850 851 Mutex::Autolock _l(mLock); 852 853 RecordThread *recordThread = checkRecordThread_l(ioHandle); 854 if (recordThread != NULL) { 855 return recordThread->getInputFramesLost(); 856 } 857 return 0; 858} 859 860status_t AudioFlinger::setVoiceVolume(float value) 861{ 862 status_t ret = initCheck(); 863 if (ret != NO_ERROR) { 864 return ret; 865 } 866 867 // check calling permissions 868 if (!settingsAllowed()) { 869 return PERMISSION_DENIED; 870 } 871 872 AutoMutex lock(mHardwareLock); 873 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 874 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 875 mHardwareStatus = AUDIO_HW_IDLE; 876 877 return ret; 878} 879 880status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 881{ 882 status_t status; 883 884 Mutex::Autolock _l(mLock); 885 886 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 887 if (playbackThread != NULL) { 888 return playbackThread->getRenderPosition(halFrames, dspFrames); 889 } 890 891 return BAD_VALUE; 892} 893 894void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 895{ 896 897 Mutex::Autolock _l(mLock); 898 899 int pid = IPCThreadState::self()->getCallingPid(); 900 if (mNotificationClients.indexOfKey(pid) < 0) { 901 sp<NotificationClient> notificationClient = new NotificationClient(this, 902 client, 903 pid); 904 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 905 906 mNotificationClients.add(pid, notificationClient); 907 908 sp<IBinder> binder = client->asBinder(); 909 binder->linkToDeath(notificationClient); 910 911 // the config change is always sent from playback or record threads to avoid deadlock 912 // with AudioSystem::gLock 913 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 914 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 915 } 916 917 for (size_t i = 0; i < mRecordThreads.size(); i++) { 918 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 919 } 920 } 921} 922 923void AudioFlinger::removeNotificationClient(pid_t pid) 924{ 925 Mutex::Autolock _l(mLock); 926 927 int index = mNotificationClients.indexOfKey(pid); 928 if (index >= 0) { 929 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 930 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 931 mNotificationClients.removeItem(pid); 932 } 933 934 ALOGV("%d died, releasing its sessions", pid); 935 int num = mAudioSessionRefs.size(); 936 bool removed = false; 937 for (int i = 0; i< num; i++) { 938 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 939 ALOGV(" pid %d @ %d", ref->pid, i); 940 if (ref->pid == pid) { 941 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 942 mAudioSessionRefs.removeAt(i); 943 delete ref; 944 removed = true; 945 i--; 946 num--; 947 } 948 } 949 if (removed) { 950 purgeStaleEffects_l(); 951 } 952} 953 954// audioConfigChanged_l() must be called with AudioFlinger::mLock held 955void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 956{ 957 size_t size = mNotificationClients.size(); 958 for (size_t i = 0; i < size; i++) { 959 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 960 } 961} 962 963// removeClient_l() must be called with AudioFlinger::mLock held 964void AudioFlinger::removeClient_l(pid_t pid) 965{ 966 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 967 mClients.removeItem(pid); 968} 969 970 971// ---------------------------------------------------------------------------- 972 973AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 974 : Thread(false), 975 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 976 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), 977 mDevice(device) 978{ 979 mDeathRecipient = new PMDeathRecipient(this); 980} 981 982AudioFlinger::ThreadBase::~ThreadBase() 983{ 984 mParamCond.broadcast(); 985 mNewParameters.clear(); 986 // do not lock the mutex in destructor 987 releaseWakeLock_l(); 988 if (mPowerManager != 0) { 989 sp<IBinder> binder = mPowerManager->asBinder(); 990 binder->unlinkToDeath(mDeathRecipient); 991 } 992} 993 994void AudioFlinger::ThreadBase::exit() 995{ 996 // keep a strong ref on ourself so that we wont get 997 // destroyed in the middle of requestExitAndWait() 998 sp <ThreadBase> strongMe = this; 999 1000 ALOGV("ThreadBase::exit"); 1001 { 1002 AutoMutex lock(&mLock); 1003 mExiting = true; 1004 requestExit(); 1005 mWaitWorkCV.signal(); 1006 } 1007 requestExitAndWait(); 1008} 1009 1010uint32_t AudioFlinger::ThreadBase::sampleRate() const 1011{ 1012 return mSampleRate; 1013} 1014 1015int AudioFlinger::ThreadBase::channelCount() const 1016{ 1017 return (int)mChannelCount; 1018} 1019 1020uint32_t AudioFlinger::ThreadBase::format() const 1021{ 1022 return mFormat; 1023} 1024 1025size_t AudioFlinger::ThreadBase::frameCount() const 1026{ 1027 return mFrameCount; 1028} 1029 1030status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1031{ 1032 status_t status; 1033 1034 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1035 Mutex::Autolock _l(mLock); 1036 1037 mNewParameters.add(keyValuePairs); 1038 mWaitWorkCV.signal(); 1039 // wait condition with timeout in case the thread loop has exited 1040 // before the request could be processed 1041 if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) { 1042 status = mParamStatus; 1043 mWaitWorkCV.signal(); 1044 } else { 1045 status = TIMED_OUT; 1046 } 1047 return status; 1048} 1049 1050void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1051{ 1052 Mutex::Autolock _l(mLock); 1053 sendConfigEvent_l(event, param); 1054} 1055 1056// sendConfigEvent_l() must be called with ThreadBase::mLock held 1057void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1058{ 1059 ConfigEvent *configEvent = new ConfigEvent(); 1060 configEvent->mEvent = event; 1061 configEvent->mParam = param; 1062 mConfigEvents.add(configEvent); 1063 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1064 mWaitWorkCV.signal(); 1065} 1066 1067void AudioFlinger::ThreadBase::processConfigEvents() 1068{ 1069 mLock.lock(); 1070 while(!mConfigEvents.isEmpty()) { 1071 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1072 ConfigEvent *configEvent = mConfigEvents[0]; 1073 mConfigEvents.removeAt(0); 1074 // release mLock before locking AudioFlinger mLock: lock order is always 1075 // AudioFlinger then ThreadBase to avoid cross deadlock 1076 mLock.unlock(); 1077 mAudioFlinger->mLock.lock(); 1078 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 1079 mAudioFlinger->mLock.unlock(); 1080 delete configEvent; 1081 mLock.lock(); 1082 } 1083 mLock.unlock(); 1084} 1085 1086status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1087{ 1088 const size_t SIZE = 256; 1089 char buffer[SIZE]; 1090 String8 result; 1091 1092 bool locked = tryLock(mLock); 1093 if (!locked) { 1094 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1095 write(fd, buffer, strlen(buffer)); 1096 } 1097 1098 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1099 result.append(buffer); 1100 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1101 result.append(buffer); 1102 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1103 result.append(buffer); 1104 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1105 result.append(buffer); 1106 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1107 result.append(buffer); 1108 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1111 result.append(buffer); 1112 1113 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1114 result.append(buffer); 1115 result.append(" Index Command"); 1116 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1117 snprintf(buffer, SIZE, "\n %02d ", i); 1118 result.append(buffer); 1119 result.append(mNewParameters[i]); 1120 } 1121 1122 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1123 result.append(buffer); 1124 snprintf(buffer, SIZE, " Index event param\n"); 1125 result.append(buffer); 1126 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1127 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1128 result.append(buffer); 1129 } 1130 result.append("\n"); 1131 1132 write(fd, result.string(), result.size()); 1133 1134 if (locked) { 1135 mLock.unlock(); 1136 } 1137 return NO_ERROR; 1138} 1139 1140status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1141{ 1142 const size_t SIZE = 256; 1143 char buffer[SIZE]; 1144 String8 result; 1145 1146 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1147 write(fd, buffer, strlen(buffer)); 1148 1149 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1150 sp<EffectChain> chain = mEffectChains[i]; 1151 if (chain != 0) { 1152 chain->dump(fd, args); 1153 } 1154 } 1155 return NO_ERROR; 1156} 1157 1158void AudioFlinger::ThreadBase::acquireWakeLock() 1159{ 1160 Mutex::Autolock _l(mLock); 1161 acquireWakeLock_l(); 1162} 1163 1164void AudioFlinger::ThreadBase::acquireWakeLock_l() 1165{ 1166 if (mPowerManager == 0) { 1167 // use checkService() to avoid blocking if power service is not up yet 1168 sp<IBinder> binder = 1169 defaultServiceManager()->checkService(String16("power")); 1170 if (binder == 0) { 1171 LOGW("Thread %s cannot connect to the power manager service", mName); 1172 } else { 1173 mPowerManager = interface_cast<IPowerManager>(binder); 1174 binder->linkToDeath(mDeathRecipient); 1175 } 1176 } 1177 if (mPowerManager != 0) { 1178 sp<IBinder> binder = new BBinder(); 1179 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1180 binder, 1181 String16(mName)); 1182 if (status == NO_ERROR) { 1183 mWakeLockToken = binder; 1184 } 1185 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1186 } 1187} 1188 1189void AudioFlinger::ThreadBase::releaseWakeLock() 1190{ 1191 Mutex::Autolock _l(mLock); 1192 releaseWakeLock_l(); 1193} 1194 1195void AudioFlinger::ThreadBase::releaseWakeLock_l() 1196{ 1197 if (mWakeLockToken != 0) { 1198 ALOGV("releaseWakeLock_l() %s", mName); 1199 if (mPowerManager != 0) { 1200 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1201 } 1202 mWakeLockToken.clear(); 1203 } 1204} 1205 1206void AudioFlinger::ThreadBase::clearPowerManager() 1207{ 1208 Mutex::Autolock _l(mLock); 1209 releaseWakeLock_l(); 1210 mPowerManager.clear(); 1211} 1212 1213void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1214{ 1215 sp<ThreadBase> thread = mThread.promote(); 1216 if (thread != 0) { 1217 thread->clearPowerManager(); 1218 } 1219 LOGW("power manager service died !!!"); 1220} 1221 1222void AudioFlinger::ThreadBase::setEffectSuspended( 1223 const effect_uuid_t *type, bool suspend, int sessionId) 1224{ 1225 Mutex::Autolock _l(mLock); 1226 setEffectSuspended_l(type, suspend, sessionId); 1227} 1228 1229void AudioFlinger::ThreadBase::setEffectSuspended_l( 1230 const effect_uuid_t *type, bool suspend, int sessionId) 1231{ 1232 sp<EffectChain> chain; 1233 chain = getEffectChain_l(sessionId); 1234 if (chain != 0) { 1235 if (type != NULL) { 1236 chain->setEffectSuspended_l(type, suspend); 1237 } else { 1238 chain->setEffectSuspendedAll_l(suspend); 1239 } 1240 } 1241 1242 updateSuspendedSessions_l(type, suspend, sessionId); 1243} 1244 1245void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1246{ 1247 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1248 if (index < 0) { 1249 return; 1250 } 1251 1252 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1253 mSuspendedSessions.editValueAt(index); 1254 1255 for (size_t i = 0; i < sessionEffects.size(); i++) { 1256 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1257 for (int j = 0; j < desc->mRefCount; j++) { 1258 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1259 chain->setEffectSuspendedAll_l(true); 1260 } else { 1261 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1262 desc->mType.timeLow); 1263 chain->setEffectSuspended_l(&desc->mType, true); 1264 } 1265 } 1266 } 1267} 1268 1269void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1270 bool suspend, 1271 int sessionId) 1272{ 1273 int index = mSuspendedSessions.indexOfKey(sessionId); 1274 1275 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1276 1277 if (suspend) { 1278 if (index >= 0) { 1279 sessionEffects = mSuspendedSessions.editValueAt(index); 1280 } else { 1281 mSuspendedSessions.add(sessionId, sessionEffects); 1282 } 1283 } else { 1284 if (index < 0) { 1285 return; 1286 } 1287 sessionEffects = mSuspendedSessions.editValueAt(index); 1288 } 1289 1290 1291 int key = EffectChain::kKeyForSuspendAll; 1292 if (type != NULL) { 1293 key = type->timeLow; 1294 } 1295 index = sessionEffects.indexOfKey(key); 1296 1297 sp <SuspendedSessionDesc> desc; 1298 if (suspend) { 1299 if (index >= 0) { 1300 desc = sessionEffects.valueAt(index); 1301 } else { 1302 desc = new SuspendedSessionDesc(); 1303 if (type != NULL) { 1304 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1305 } 1306 sessionEffects.add(key, desc); 1307 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1308 } 1309 desc->mRefCount++; 1310 } else { 1311 if (index < 0) { 1312 return; 1313 } 1314 desc = sessionEffects.valueAt(index); 1315 if (--desc->mRefCount == 0) { 1316 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1317 sessionEffects.removeItemsAt(index); 1318 if (sessionEffects.isEmpty()) { 1319 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1320 sessionId); 1321 mSuspendedSessions.removeItem(sessionId); 1322 } 1323 } 1324 } 1325 if (!sessionEffects.isEmpty()) { 1326 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1327 } 1328} 1329 1330void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1331 bool enabled, 1332 int sessionId) 1333{ 1334 Mutex::Autolock _l(mLock); 1335 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1336} 1337 1338void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1339 bool enabled, 1340 int sessionId) 1341{ 1342 if (mType != RECORD) { 1343 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1344 // another session. This gives the priority to well behaved effect control panels 1345 // and applications not using global effects. 1346 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1347 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1348 } 1349 } 1350 1351 sp<EffectChain> chain = getEffectChain_l(sessionId); 1352 if (chain != 0) { 1353 chain->checkSuspendOnEffectEnabled(effect, enabled); 1354 } 1355} 1356 1357// ---------------------------------------------------------------------------- 1358 1359AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1360 AudioStreamOut* output, 1361 int id, 1362 uint32_t device) 1363 : ThreadBase(audioFlinger, id, device), 1364 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1365 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1366{ 1367 snprintf(mName, kNameLength, "AudioOut_%d", id); 1368 1369 readOutputParameters(); 1370 1371 mMasterVolume = mAudioFlinger->masterVolume(); 1372 mMasterMute = mAudioFlinger->masterMute(); 1373 1374 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1375 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1376 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1377 mStreamTypes[stream].valid = true; 1378 } 1379} 1380 1381AudioFlinger::PlaybackThread::~PlaybackThread() 1382{ 1383 delete [] mMixBuffer; 1384} 1385 1386status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1387{ 1388 dumpInternals(fd, args); 1389 dumpTracks(fd, args); 1390 dumpEffectChains(fd, args); 1391 return NO_ERROR; 1392} 1393 1394status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1395{ 1396 const size_t SIZE = 256; 1397 char buffer[SIZE]; 1398 String8 result; 1399 1400 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1401 result.append(buffer); 1402 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1403 for (size_t i = 0; i < mTracks.size(); ++i) { 1404 sp<Track> track = mTracks[i]; 1405 if (track != 0) { 1406 track->dump(buffer, SIZE); 1407 result.append(buffer); 1408 } 1409 } 1410 1411 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1412 result.append(buffer); 1413 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1414 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1415 wp<Track> wTrack = mActiveTracks[i]; 1416 if (wTrack != 0) { 1417 sp<Track> track = wTrack.promote(); 1418 if (track != 0) { 1419 track->dump(buffer, SIZE); 1420 result.append(buffer); 1421 } 1422 } 1423 } 1424 write(fd, result.string(), result.size()); 1425 return NO_ERROR; 1426} 1427 1428status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1429{ 1430 const size_t SIZE = 256; 1431 char buffer[SIZE]; 1432 String8 result; 1433 1434 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1435 result.append(buffer); 1436 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1437 result.append(buffer); 1438 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1439 result.append(buffer); 1440 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1441 result.append(buffer); 1442 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1443 result.append(buffer); 1444 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1445 result.append(buffer); 1446 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1447 result.append(buffer); 1448 write(fd, result.string(), result.size()); 1449 1450 dumpBase(fd, args); 1451 1452 return NO_ERROR; 1453} 1454 1455// Thread virtuals 1456status_t AudioFlinger::PlaybackThread::readyToRun() 1457{ 1458 status_t status = initCheck(); 1459 if (status == NO_ERROR) { 1460 LOGI("AudioFlinger's thread %p ready to run", this); 1461 } else { 1462 LOGE("No working audio driver found."); 1463 } 1464 return status; 1465} 1466 1467void AudioFlinger::PlaybackThread::onFirstRef() 1468{ 1469 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1470} 1471 1472// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1473sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1474 const sp<AudioFlinger::Client>& client, 1475 int streamType, 1476 uint32_t sampleRate, 1477 uint32_t format, 1478 uint32_t channelMask, 1479 int frameCount, 1480 const sp<IMemory>& sharedBuffer, 1481 int sessionId, 1482 status_t *status) 1483{ 1484 sp<Track> track; 1485 status_t lStatus; 1486 1487 if (mType == DIRECT) { 1488 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1489 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1490 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1491 "for output %p with format %d", 1492 sampleRate, format, channelMask, mOutput, mFormat); 1493 lStatus = BAD_VALUE; 1494 goto Exit; 1495 } 1496 } 1497 } else { 1498 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1499 if (sampleRate > mSampleRate*2) { 1500 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1501 lStatus = BAD_VALUE; 1502 goto Exit; 1503 } 1504 } 1505 1506 lStatus = initCheck(); 1507 if (lStatus != NO_ERROR) { 1508 LOGE("Audio driver not initialized."); 1509 goto Exit; 1510 } 1511 1512 { // scope for mLock 1513 Mutex::Autolock _l(mLock); 1514 1515 // all tracks in same audio session must share the same routing strategy otherwise 1516 // conflicts will happen when tracks are moved from one output to another by audio policy 1517 // manager 1518 uint32_t strategy = 1519 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1520 for (size_t i = 0; i < mTracks.size(); ++i) { 1521 sp<Track> t = mTracks[i]; 1522 if (t != 0) { 1523 if (sessionId == t->sessionId() && 1524 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1525 lStatus = BAD_VALUE; 1526 goto Exit; 1527 } 1528 } 1529 } 1530 1531 track = new Track(this, client, streamType, sampleRate, format, 1532 channelMask, frameCount, sharedBuffer, sessionId); 1533 if (track->getCblk() == NULL || track->name() < 0) { 1534 lStatus = NO_MEMORY; 1535 goto Exit; 1536 } 1537 mTracks.add(track); 1538 1539 sp<EffectChain> chain = getEffectChain_l(sessionId); 1540 if (chain != 0) { 1541 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1542 track->setMainBuffer(chain->inBuffer()); 1543 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1544 chain->incTrackCnt(); 1545 } 1546 1547 // invalidate track immediately if the stream type was moved to another thread since 1548 // createTrack() was called by the client process. 1549 if (!mStreamTypes[streamType].valid) { 1550 LOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1551 this, streamType); 1552 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1553 } 1554 } 1555 lStatus = NO_ERROR; 1556 1557Exit: 1558 if(status) { 1559 *status = lStatus; 1560 } 1561 return track; 1562} 1563 1564uint32_t AudioFlinger::PlaybackThread::latency() const 1565{ 1566 Mutex::Autolock _l(mLock); 1567 if (initCheck() == NO_ERROR) { 1568 return mOutput->stream->get_latency(mOutput->stream); 1569 } else { 1570 return 0; 1571 } 1572} 1573 1574status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1575{ 1576 mMasterVolume = value; 1577 return NO_ERROR; 1578} 1579 1580status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1581{ 1582 mMasterMute = muted; 1583 return NO_ERROR; 1584} 1585 1586float AudioFlinger::PlaybackThread::masterVolume() const 1587{ 1588 return mMasterVolume; 1589} 1590 1591bool AudioFlinger::PlaybackThread::masterMute() const 1592{ 1593 return mMasterMute; 1594} 1595 1596status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1597{ 1598 mStreamTypes[stream].volume = value; 1599 return NO_ERROR; 1600} 1601 1602status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1603{ 1604 mStreamTypes[stream].mute = muted; 1605 return NO_ERROR; 1606} 1607 1608float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1609{ 1610 return mStreamTypes[stream].volume; 1611} 1612 1613bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1614{ 1615 return mStreamTypes[stream].mute; 1616} 1617 1618// addTrack_l() must be called with ThreadBase::mLock held 1619status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1620{ 1621 status_t status = ALREADY_EXISTS; 1622 1623 // set retry count for buffer fill 1624 track->mRetryCount = kMaxTrackStartupRetries; 1625 if (mActiveTracks.indexOf(track) < 0) { 1626 // the track is newly added, make sure it fills up all its 1627 // buffers before playing. This is to ensure the client will 1628 // effectively get the latency it requested. 1629 track->mFillingUpStatus = Track::FS_FILLING; 1630 track->mResetDone = false; 1631 mActiveTracks.add(track); 1632 if (track->mainBuffer() != mMixBuffer) { 1633 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1634 if (chain != 0) { 1635 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1636 chain->incActiveTrackCnt(); 1637 } 1638 } 1639 1640 status = NO_ERROR; 1641 } 1642 1643 ALOGV("mWaitWorkCV.broadcast"); 1644 mWaitWorkCV.broadcast(); 1645 1646 return status; 1647} 1648 1649// destroyTrack_l() must be called with ThreadBase::mLock held 1650void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1651{ 1652 track->mState = TrackBase::TERMINATED; 1653 if (mActiveTracks.indexOf(track) < 0) { 1654 removeTrack_l(track); 1655 } 1656} 1657 1658void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1659{ 1660 mTracks.remove(track); 1661 deleteTrackName_l(track->name()); 1662 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1663 if (chain != 0) { 1664 chain->decTrackCnt(); 1665 } 1666} 1667 1668String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1669{ 1670 String8 out_s8 = String8(""); 1671 char *s; 1672 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() != NO_ERROR) { 1675 return out_s8; 1676 } 1677 1678 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1679 out_s8 = String8(s); 1680 free(s); 1681 return out_s8; 1682} 1683 1684// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1685void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1686 AudioSystem::OutputDescriptor desc; 1687 void *param2 = 0; 1688 1689 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1690 1691 switch (event) { 1692 case AudioSystem::OUTPUT_OPENED: 1693 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1694 desc.channels = mChannelMask; 1695 desc.samplingRate = mSampleRate; 1696 desc.format = mFormat; 1697 desc.frameCount = mFrameCount; 1698 desc.latency = latency(); 1699 param2 = &desc; 1700 break; 1701 1702 case AudioSystem::STREAM_CONFIG_CHANGED: 1703 param2 = ¶m; 1704 case AudioSystem::OUTPUT_CLOSED: 1705 default: 1706 break; 1707 } 1708 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1709} 1710 1711void AudioFlinger::PlaybackThread::readOutputParameters() 1712{ 1713 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1714 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1715 mChannelCount = (uint16_t)popcount(mChannelMask); 1716 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1717 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1718 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1719 1720 // FIXME - Current mixer implementation only supports stereo output: Always 1721 // Allocate a stereo buffer even if HW output is mono. 1722 if (mMixBuffer != NULL) delete[] mMixBuffer; 1723 mMixBuffer = new int16_t[mFrameCount * 2]; 1724 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1725 1726 // force reconfiguration of effect chains and engines to take new buffer size and audio 1727 // parameters into account 1728 // Note that mLock is not held when readOutputParameters() is called from the constructor 1729 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1730 // matter. 1731 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1732 Vector< sp<EffectChain> > effectChains = mEffectChains; 1733 for (size_t i = 0; i < effectChains.size(); i ++) { 1734 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1735 } 1736} 1737 1738status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1739{ 1740 if (halFrames == 0 || dspFrames == 0) { 1741 return BAD_VALUE; 1742 } 1743 Mutex::Autolock _l(mLock); 1744 if (initCheck() != NO_ERROR) { 1745 return INVALID_OPERATION; 1746 } 1747 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1748 1749 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1750} 1751 1752uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1753{ 1754 Mutex::Autolock _l(mLock); 1755 uint32_t result = 0; 1756 if (getEffectChain_l(sessionId) != 0) { 1757 result = EFFECT_SESSION; 1758 } 1759 1760 for (size_t i = 0; i < mTracks.size(); ++i) { 1761 sp<Track> track = mTracks[i]; 1762 if (sessionId == track->sessionId() && 1763 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1764 result |= TRACK_SESSION; 1765 break; 1766 } 1767 } 1768 1769 return result; 1770} 1771 1772uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1773{ 1774 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1775 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1776 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1777 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1778 } 1779 for (size_t i = 0; i < mTracks.size(); i++) { 1780 sp<Track> track = mTracks[i]; 1781 if (sessionId == track->sessionId() && 1782 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1783 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1784 } 1785 } 1786 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1787} 1788 1789 1790AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() 1791{ 1792 Mutex::Autolock _l(mLock); 1793 return mOutput; 1794} 1795 1796AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1797{ 1798 Mutex::Autolock _l(mLock); 1799 AudioStreamOut *output = mOutput; 1800 mOutput = NULL; 1801 return output; 1802} 1803 1804// this method must always be called either with ThreadBase mLock held or inside the thread loop 1805audio_stream_t* AudioFlinger::PlaybackThread::stream() 1806{ 1807 if (mOutput == NULL) { 1808 return NULL; 1809 } 1810 return &mOutput->stream->common; 1811} 1812 1813// ---------------------------------------------------------------------------- 1814 1815AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1816 : PlaybackThread(audioFlinger, output, id, device), 1817 mAudioMixer(0) 1818{ 1819 mType = ThreadBase::MIXER; 1820 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1821 1822 // FIXME - Current mixer implementation only supports stereo output 1823 if (mChannelCount == 1) { 1824 LOGE("Invalid audio hardware channel count"); 1825 } 1826} 1827 1828AudioFlinger::MixerThread::~MixerThread() 1829{ 1830 delete mAudioMixer; 1831} 1832 1833bool AudioFlinger::MixerThread::threadLoop() 1834{ 1835 Vector< sp<Track> > tracksToRemove; 1836 uint32_t mixerStatus = MIXER_IDLE; 1837 nsecs_t standbyTime = systemTime(); 1838 size_t mixBufferSize = mFrameCount * mFrameSize; 1839 // FIXME: Relaxed timing because of a certain device that can't meet latency 1840 // Should be reduced to 2x after the vendor fixes the driver issue 1841 // increase threshold again due to low power audio mode. The way this warning threshold is 1842 // calculated and its usefulness should be reconsidered anyway. 1843 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1844 nsecs_t lastWarning = 0; 1845 bool longStandbyExit = false; 1846 uint32_t activeSleepTime = activeSleepTimeUs(); 1847 uint32_t idleSleepTime = idleSleepTimeUs(); 1848 uint32_t sleepTime = idleSleepTime; 1849 Vector< sp<EffectChain> > effectChains; 1850#ifdef DEBUG_CPU_USAGE 1851 ThreadCpuUsage cpu; 1852 const CentralTendencyStatistics& stats = cpu.statistics(); 1853#endif 1854 1855 acquireWakeLock(); 1856 1857 while (!exitPending()) 1858 { 1859#ifdef DEBUG_CPU_USAGE 1860 cpu.sampleAndEnable(); 1861 unsigned n = stats.n(); 1862 // cpu.elapsed() is expensive, so don't call it every loop 1863 if ((n & 127) == 1) { 1864 long long elapsed = cpu.elapsed(); 1865 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1866 double perLoop = elapsed / (double) n; 1867 double perLoop100 = perLoop * 0.01; 1868 double mean = stats.mean(); 1869 double stddev = stats.stddev(); 1870 double minimum = stats.minimum(); 1871 double maximum = stats.maximum(); 1872 cpu.resetStatistics(); 1873 LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1874 elapsed * .000000001, n, perLoop * .000001, 1875 mean * .001, 1876 stddev * .001, 1877 minimum * .001, 1878 maximum * .001, 1879 mean / perLoop100, 1880 stddev / perLoop100, 1881 minimum / perLoop100, 1882 maximum / perLoop100); 1883 } 1884 } 1885#endif 1886 processConfigEvents(); 1887 1888 mixerStatus = MIXER_IDLE; 1889 { // scope for mLock 1890 1891 Mutex::Autolock _l(mLock); 1892 1893 if (checkForNewParameters_l()) { 1894 mixBufferSize = mFrameCount * mFrameSize; 1895 // FIXME: Relaxed timing because of a certain device that can't meet latency 1896 // Should be reduced to 2x after the vendor fixes the driver issue 1897 // increase threshold again due to low power audio mode. The way this warning 1898 // threshold is calculated and its usefulness should be reconsidered anyway. 1899 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1900 activeSleepTime = activeSleepTimeUs(); 1901 idleSleepTime = idleSleepTimeUs(); 1902 } 1903 1904 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1905 1906 // put audio hardware into standby after short delay 1907 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1908 mSuspended) { 1909 if (!mStandby) { 1910 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1911 mOutput->stream->common.standby(&mOutput->stream->common); 1912 mStandby = true; 1913 mBytesWritten = 0; 1914 } 1915 1916 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1917 // we're about to wait, flush the binder command buffer 1918 IPCThreadState::self()->flushCommands(); 1919 1920 if (exitPending()) break; 1921 1922 releaseWakeLock_l(); 1923 // wait until we have something to do... 1924 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1925 mWaitWorkCV.wait(mLock); 1926 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1927 acquireWakeLock_l(); 1928 1929 if (mMasterMute == false) { 1930 char value[PROPERTY_VALUE_MAX]; 1931 property_get("ro.audio.silent", value, "0"); 1932 if (atoi(value)) { 1933 LOGD("Silence is golden"); 1934 setMasterMute(true); 1935 } 1936 } 1937 1938 standbyTime = systemTime() + kStandbyTimeInNsecs; 1939 sleepTime = idleSleepTime; 1940 continue; 1941 } 1942 } 1943 1944 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1945 1946 // prevent any changes in effect chain list and in each effect chain 1947 // during mixing and effect process as the audio buffers could be deleted 1948 // or modified if an effect is created or deleted 1949 lockEffectChains_l(effectChains); 1950 } 1951 1952 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1953 // mix buffers... 1954 mAudioMixer->process(); 1955 sleepTime = 0; 1956 standbyTime = systemTime() + kStandbyTimeInNsecs; 1957 //TODO: delay standby when effects have a tail 1958 } else { 1959 // If no tracks are ready, sleep once for the duration of an output 1960 // buffer size, then write 0s to the output 1961 if (sleepTime == 0) { 1962 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1963 sleepTime = activeSleepTime; 1964 } else { 1965 sleepTime = idleSleepTime; 1966 } 1967 } else if (mBytesWritten != 0 || 1968 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1969 memset (mMixBuffer, 0, mixBufferSize); 1970 sleepTime = 0; 1971 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1972 } 1973 // TODO add standby time extension fct of effect tail 1974 } 1975 1976 if (mSuspended) { 1977 sleepTime = suspendSleepTimeUs(); 1978 } 1979 // sleepTime == 0 means we must write to audio hardware 1980 if (sleepTime == 0) { 1981 for (size_t i = 0; i < effectChains.size(); i ++) { 1982 effectChains[i]->process_l(); 1983 } 1984 // enable changes in effect chain 1985 unlockEffectChains(effectChains); 1986 mLastWriteTime = systemTime(); 1987 mInWrite = true; 1988 mBytesWritten += mixBufferSize; 1989 1990 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1991 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1992 mNumWrites++; 1993 mInWrite = false; 1994 nsecs_t now = systemTime(); 1995 nsecs_t delta = now - mLastWriteTime; 1996 if (!mStandby && delta > maxPeriod) { 1997 mNumDelayedWrites++; 1998 if ((now - lastWarning) > kWarningThrottle) { 1999 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2000 ns2ms(delta), mNumDelayedWrites, this); 2001 lastWarning = now; 2002 } 2003 if (mStandby) { 2004 longStandbyExit = true; 2005 } 2006 } 2007 mStandby = false; 2008 } else { 2009 // enable changes in effect chain 2010 unlockEffectChains(effectChains); 2011 usleep(sleepTime); 2012 } 2013 2014 // finally let go of all our tracks, without the lock held 2015 // since we can't guarantee the destructors won't acquire that 2016 // same lock. 2017 tracksToRemove.clear(); 2018 2019 // Effect chains will be actually deleted here if they were removed from 2020 // mEffectChains list during mixing or effects processing 2021 effectChains.clear(); 2022 } 2023 2024 if (!mStandby) { 2025 mOutput->stream->common.standby(&mOutput->stream->common); 2026 } 2027 2028 releaseWakeLock(); 2029 2030 ALOGV("MixerThread %p exiting", this); 2031 return false; 2032} 2033 2034// prepareTracks_l() must be called with ThreadBase::mLock held 2035uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2036{ 2037 2038 uint32_t mixerStatus = MIXER_IDLE; 2039 // find out which tracks need to be processed 2040 size_t count = activeTracks.size(); 2041 size_t mixedTracks = 0; 2042 size_t tracksWithEffect = 0; 2043 2044 float masterVolume = mMasterVolume; 2045 bool masterMute = mMasterMute; 2046 2047 if (masterMute) { 2048 masterVolume = 0; 2049 } 2050 // Delegate master volume control to effect in output mix effect chain if needed 2051 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2052 if (chain != 0) { 2053 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2054 chain->setVolume_l(&v, &v); 2055 masterVolume = (float)((v + (1 << 23)) >> 24); 2056 chain.clear(); 2057 } 2058 2059 for (size_t i=0 ; i<count ; i++) { 2060 sp<Track> t = activeTracks[i].promote(); 2061 if (t == 0) continue; 2062 2063 Track* const track = t.get(); 2064 audio_track_cblk_t* cblk = track->cblk(); 2065 2066 // The first time a track is added we wait 2067 // for all its buffers to be filled before processing it 2068 mAudioMixer->setActiveTrack(track->name()); 2069 // make sure that we have enough frames to mix one full buffer 2070 uint32_t minFrames = 1; 2071 if (!track->isStopped() && !track->isPausing()) { 2072 if (t->sampleRate() == (int)mSampleRate) { 2073 minFrames = mFrameCount; 2074 } else { 2075 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1; 2076 } 2077 } 2078 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2079 !track->isPaused() && !track->isTerminated()) 2080 { 2081 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 2082 2083 mixedTracks++; 2084 2085 // track->mainBuffer() != mMixBuffer means there is an effect chain 2086 // connected to the track 2087 chain.clear(); 2088 if (track->mainBuffer() != mMixBuffer) { 2089 chain = getEffectChain_l(track->sessionId()); 2090 // Delegate volume control to effect in track effect chain if needed 2091 if (chain != 0) { 2092 tracksWithEffect++; 2093 } else { 2094 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 2095 track->name(), track->sessionId()); 2096 } 2097 } 2098 2099 2100 int param = AudioMixer::VOLUME; 2101 if (track->mFillingUpStatus == Track::FS_FILLED) { 2102 // no ramp for the first volume setting 2103 track->mFillingUpStatus = Track::FS_ACTIVE; 2104 if (track->mState == TrackBase::RESUMING) { 2105 track->mState = TrackBase::ACTIVE; 2106 param = AudioMixer::RAMP_VOLUME; 2107 } 2108 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2109 } else if (cblk->server != 0) { 2110 // If the track is stopped before the first frame was mixed, 2111 // do not apply ramp 2112 param = AudioMixer::RAMP_VOLUME; 2113 } 2114 2115 // compute volume for this track 2116 uint32_t vl, vr, va; 2117 if (track->isMuted() || track->isPausing() || 2118 mStreamTypes[track->type()].mute) { 2119 vl = vr = va = 0; 2120 if (track->isPausing()) { 2121 track->setPaused(); 2122 } 2123 } else { 2124 2125 // read original volumes with volume control 2126 float typeVolume = mStreamTypes[track->type()].volume; 2127 float v = masterVolume * typeVolume; 2128 vl = (uint32_t)(v * cblk->volume[0]) << 12; 2129 vr = (uint32_t)(v * cblk->volume[1]) << 12; 2130 2131 va = (uint32_t)(v * cblk->sendLevel); 2132 } 2133 // Delegate volume control to effect in track effect chain if needed 2134 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2135 // Do not ramp volume if volume is controlled by effect 2136 param = AudioMixer::VOLUME; 2137 track->mHasVolumeController = true; 2138 } else { 2139 // force no volume ramp when volume controller was just disabled or removed 2140 // from effect chain to avoid volume spike 2141 if (track->mHasVolumeController) { 2142 param = AudioMixer::VOLUME; 2143 } 2144 track->mHasVolumeController = false; 2145 } 2146 2147 // Convert volumes from 8.24 to 4.12 format 2148 int16_t left, right, aux; 2149 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2150 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2151 left = int16_t(v_clamped); 2152 v_clamped = (vr + (1 << 11)) >> 12; 2153 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2154 right = int16_t(v_clamped); 2155 2156 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2157 aux = int16_t(va); 2158 2159 // XXX: these things DON'T need to be done each time 2160 mAudioMixer->setBufferProvider(track); 2161 mAudioMixer->enable(AudioMixer::MIXING); 2162 2163 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 2164 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 2165 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 2166 mAudioMixer->setParameter( 2167 AudioMixer::TRACK, 2168 AudioMixer::FORMAT, (void *)track->format()); 2169 mAudioMixer->setParameter( 2170 AudioMixer::TRACK, 2171 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2172 mAudioMixer->setParameter( 2173 AudioMixer::RESAMPLE, 2174 AudioMixer::SAMPLE_RATE, 2175 (void *)(cblk->sampleRate)); 2176 mAudioMixer->setParameter( 2177 AudioMixer::TRACK, 2178 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2179 mAudioMixer->setParameter( 2180 AudioMixer::TRACK, 2181 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2182 2183 // reset retry count 2184 track->mRetryCount = kMaxTrackRetries; 2185 mixerStatus = MIXER_TRACKS_READY; 2186 } else { 2187 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 2188 if (track->isStopped()) { 2189 track->reset(); 2190 } 2191 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2192 // We have consumed all the buffers of this track. 2193 // Remove it from the list of active tracks. 2194 tracksToRemove->add(track); 2195 } else { 2196 // No buffers for this track. Give it a few chances to 2197 // fill a buffer, then remove it from active list. 2198 if (--(track->mRetryCount) <= 0) { 2199 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 2200 tracksToRemove->add(track); 2201 // indicate to client process that the track was disabled because of underrun 2202 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2203 } else if (mixerStatus != MIXER_TRACKS_READY) { 2204 mixerStatus = MIXER_TRACKS_ENABLED; 2205 } 2206 } 2207 mAudioMixer->disable(AudioMixer::MIXING); 2208 } 2209 } 2210 2211 // remove all the tracks that need to be... 2212 count = tracksToRemove->size(); 2213 if (UNLIKELY(count)) { 2214 for (size_t i=0 ; i<count ; i++) { 2215 const sp<Track>& track = tracksToRemove->itemAt(i); 2216 mActiveTracks.remove(track); 2217 if (track->mainBuffer() != mMixBuffer) { 2218 chain = getEffectChain_l(track->sessionId()); 2219 if (chain != 0) { 2220 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2221 chain->decActiveTrackCnt(); 2222 } 2223 } 2224 if (track->isTerminated()) { 2225 removeTrack_l(track); 2226 } 2227 } 2228 } 2229 2230 // mix buffer must be cleared if all tracks are connected to an 2231 // effect chain as in this case the mixer will not write to 2232 // mix buffer and track effects will accumulate into it 2233 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2234 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2235 } 2236 2237 return mixerStatus; 2238} 2239 2240void AudioFlinger::MixerThread::invalidateTracks(int streamType) 2241{ 2242 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2243 this, streamType, mTracks.size()); 2244 Mutex::Autolock _l(mLock); 2245 2246 size_t size = mTracks.size(); 2247 for (size_t i = 0; i < size; i++) { 2248 sp<Track> t = mTracks[i]; 2249 if (t->type() == streamType) { 2250 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2251 t->mCblk->cv.signal(); 2252 } 2253 } 2254} 2255 2256void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid) 2257{ 2258 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2259 this, streamType, valid); 2260 Mutex::Autolock _l(mLock); 2261 2262 mStreamTypes[streamType].valid = valid; 2263} 2264 2265// getTrackName_l() must be called with ThreadBase::mLock held 2266int AudioFlinger::MixerThread::getTrackName_l() 2267{ 2268 return mAudioMixer->getTrackName(); 2269} 2270 2271// deleteTrackName_l() must be called with ThreadBase::mLock held 2272void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2273{ 2274 ALOGV("remove track (%d) and delete from mixer", name); 2275 mAudioMixer->deleteTrackName(name); 2276} 2277 2278// checkForNewParameters_l() must be called with ThreadBase::mLock held 2279bool AudioFlinger::MixerThread::checkForNewParameters_l() 2280{ 2281 bool reconfig = false; 2282 2283 while (!mNewParameters.isEmpty()) { 2284 status_t status = NO_ERROR; 2285 String8 keyValuePair = mNewParameters[0]; 2286 AudioParameter param = AudioParameter(keyValuePair); 2287 int value; 2288 2289 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2290 reconfig = true; 2291 } 2292 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2293 if (value != AUDIO_FORMAT_PCM_16_BIT) { 2294 status = BAD_VALUE; 2295 } else { 2296 reconfig = true; 2297 } 2298 } 2299 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2300 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2301 status = BAD_VALUE; 2302 } else { 2303 reconfig = true; 2304 } 2305 } 2306 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2307 // do not accept frame count changes if tracks are open as the track buffer 2308 // size depends on frame count and correct behavior would not be garantied 2309 // if frame count is changed after track creation 2310 if (!mTracks.isEmpty()) { 2311 status = INVALID_OPERATION; 2312 } else { 2313 reconfig = true; 2314 } 2315 } 2316 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2317 // when changing the audio output device, call addBatteryData to notify 2318 // the change 2319 if ((int)mDevice != value) { 2320 uint32_t params = 0; 2321 // check whether speaker is on 2322 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2323 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2324 } 2325 2326 int deviceWithoutSpeaker 2327 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2328 // check if any other device (except speaker) is on 2329 if (value & deviceWithoutSpeaker ) { 2330 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2331 } 2332 2333 if (params != 0) { 2334 addBatteryData(params); 2335 } 2336 } 2337 2338 // forward device change to effects that have requested to be 2339 // aware of attached audio device. 2340 mDevice = (uint32_t)value; 2341 for (size_t i = 0; i < mEffectChains.size(); i++) { 2342 mEffectChains[i]->setDevice_l(mDevice); 2343 } 2344 } 2345 2346 if (status == NO_ERROR) { 2347 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2348 keyValuePair.string()); 2349 if (!mStandby && status == INVALID_OPERATION) { 2350 mOutput->stream->common.standby(&mOutput->stream->common); 2351 mStandby = true; 2352 mBytesWritten = 0; 2353 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2354 keyValuePair.string()); 2355 } 2356 if (status == NO_ERROR && reconfig) { 2357 delete mAudioMixer; 2358 readOutputParameters(); 2359 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2360 for (size_t i = 0; i < mTracks.size() ; i++) { 2361 int name = getTrackName_l(); 2362 if (name < 0) break; 2363 mTracks[i]->mName = name; 2364 // limit track sample rate to 2 x new output sample rate 2365 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2366 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2367 } 2368 } 2369 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2370 } 2371 } 2372 2373 mNewParameters.removeAt(0); 2374 2375 mParamStatus = status; 2376 mParamCond.signal(); 2377 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2378 // already timed out waiting for the status and will never signal the condition. 2379 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2380 } 2381 return reconfig; 2382} 2383 2384status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2385{ 2386 const size_t SIZE = 256; 2387 char buffer[SIZE]; 2388 String8 result; 2389 2390 PlaybackThread::dumpInternals(fd, args); 2391 2392 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2393 result.append(buffer); 2394 write(fd, result.string(), result.size()); 2395 return NO_ERROR; 2396} 2397 2398uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2399{ 2400 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2401} 2402 2403uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2404{ 2405 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2406} 2407 2408uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2409{ 2410 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2411} 2412 2413// ---------------------------------------------------------------------------- 2414AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2415 : PlaybackThread(audioFlinger, output, id, device) 2416{ 2417 mType = ThreadBase::DIRECT; 2418} 2419 2420AudioFlinger::DirectOutputThread::~DirectOutputThread() 2421{ 2422} 2423 2424 2425static inline int16_t clamp16(int32_t sample) 2426{ 2427 if ((sample>>15) ^ (sample>>31)) 2428 sample = 0x7FFF ^ (sample>>31); 2429 return sample; 2430} 2431 2432static inline 2433int32_t mul(int16_t in, int16_t v) 2434{ 2435#if defined(__arm__) && !defined(__thumb__) 2436 int32_t out; 2437 asm( "smulbb %[out], %[in], %[v] \n" 2438 : [out]"=r"(out) 2439 : [in]"%r"(in), [v]"r"(v) 2440 : ); 2441 return out; 2442#else 2443 return in * int32_t(v); 2444#endif 2445} 2446 2447void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2448{ 2449 // Do not apply volume on compressed audio 2450 if (!audio_is_linear_pcm(mFormat)) { 2451 return; 2452 } 2453 2454 // convert to signed 16 bit before volume calculation 2455 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2456 size_t count = mFrameCount * mChannelCount; 2457 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2458 int16_t *dst = mMixBuffer + count-1; 2459 while(count--) { 2460 *dst-- = (int16_t)(*src--^0x80) << 8; 2461 } 2462 } 2463 2464 size_t frameCount = mFrameCount; 2465 int16_t *out = mMixBuffer; 2466 if (ramp) { 2467 if (mChannelCount == 1) { 2468 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2469 int32_t vlInc = d / (int32_t)frameCount; 2470 int32_t vl = ((int32_t)mLeftVolShort << 16); 2471 do { 2472 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2473 out++; 2474 vl += vlInc; 2475 } while (--frameCount); 2476 2477 } else { 2478 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2479 int32_t vlInc = d / (int32_t)frameCount; 2480 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2481 int32_t vrInc = d / (int32_t)frameCount; 2482 int32_t vl = ((int32_t)mLeftVolShort << 16); 2483 int32_t vr = ((int32_t)mRightVolShort << 16); 2484 do { 2485 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2486 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2487 out += 2; 2488 vl += vlInc; 2489 vr += vrInc; 2490 } while (--frameCount); 2491 } 2492 } else { 2493 if (mChannelCount == 1) { 2494 do { 2495 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2496 out++; 2497 } while (--frameCount); 2498 } else { 2499 do { 2500 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2501 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2502 out += 2; 2503 } while (--frameCount); 2504 } 2505 } 2506 2507 // convert back to unsigned 8 bit after volume calculation 2508 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2509 size_t count = mFrameCount * mChannelCount; 2510 int16_t *src = mMixBuffer; 2511 uint8_t *dst = (uint8_t *)mMixBuffer; 2512 while(count--) { 2513 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2514 } 2515 } 2516 2517 mLeftVolShort = leftVol; 2518 mRightVolShort = rightVol; 2519} 2520 2521bool AudioFlinger::DirectOutputThread::threadLoop() 2522{ 2523 uint32_t mixerStatus = MIXER_IDLE; 2524 sp<Track> trackToRemove; 2525 sp<Track> activeTrack; 2526 nsecs_t standbyTime = systemTime(); 2527 int8_t *curBuf; 2528 size_t mixBufferSize = mFrameCount*mFrameSize; 2529 uint32_t activeSleepTime = activeSleepTimeUs(); 2530 uint32_t idleSleepTime = idleSleepTimeUs(); 2531 uint32_t sleepTime = idleSleepTime; 2532 // use shorter standby delay as on normal output to release 2533 // hardware resources as soon as possible 2534 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2535 2536 acquireWakeLock(); 2537 2538 while (!exitPending()) 2539 { 2540 bool rampVolume; 2541 uint16_t leftVol; 2542 uint16_t rightVol; 2543 Vector< sp<EffectChain> > effectChains; 2544 2545 processConfigEvents(); 2546 2547 mixerStatus = MIXER_IDLE; 2548 2549 { // scope for the mLock 2550 2551 Mutex::Autolock _l(mLock); 2552 2553 if (checkForNewParameters_l()) { 2554 mixBufferSize = mFrameCount*mFrameSize; 2555 activeSleepTime = activeSleepTimeUs(); 2556 idleSleepTime = idleSleepTimeUs(); 2557 standbyDelay = microseconds(activeSleepTime*2); 2558 } 2559 2560 // put audio hardware into standby after short delay 2561 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2562 mSuspended) { 2563 // wait until we have something to do... 2564 if (!mStandby) { 2565 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2566 mOutput->stream->common.standby(&mOutput->stream->common); 2567 mStandby = true; 2568 mBytesWritten = 0; 2569 } 2570 2571 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2572 // we're about to wait, flush the binder command buffer 2573 IPCThreadState::self()->flushCommands(); 2574 2575 if (exitPending()) break; 2576 2577 releaseWakeLock_l(); 2578 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2579 mWaitWorkCV.wait(mLock); 2580 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2581 acquireWakeLock_l(); 2582 2583 if (mMasterMute == false) { 2584 char value[PROPERTY_VALUE_MAX]; 2585 property_get("ro.audio.silent", value, "0"); 2586 if (atoi(value)) { 2587 LOGD("Silence is golden"); 2588 setMasterMute(true); 2589 } 2590 } 2591 2592 standbyTime = systemTime() + standbyDelay; 2593 sleepTime = idleSleepTime; 2594 continue; 2595 } 2596 } 2597 2598 effectChains = mEffectChains; 2599 2600 // find out which tracks need to be processed 2601 if (mActiveTracks.size() != 0) { 2602 sp<Track> t = mActiveTracks[0].promote(); 2603 if (t == 0) continue; 2604 2605 Track* const track = t.get(); 2606 audio_track_cblk_t* cblk = track->cblk(); 2607 2608 // The first time a track is added we wait 2609 // for all its buffers to be filled before processing it 2610 if (cblk->framesReady() && track->isReady() && 2611 !track->isPaused() && !track->isTerminated()) 2612 { 2613 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2614 2615 if (track->mFillingUpStatus == Track::FS_FILLED) { 2616 track->mFillingUpStatus = Track::FS_ACTIVE; 2617 mLeftVolFloat = mRightVolFloat = 0; 2618 mLeftVolShort = mRightVolShort = 0; 2619 if (track->mState == TrackBase::RESUMING) { 2620 track->mState = TrackBase::ACTIVE; 2621 rampVolume = true; 2622 } 2623 } else if (cblk->server != 0) { 2624 // If the track is stopped before the first frame was mixed, 2625 // do not apply ramp 2626 rampVolume = true; 2627 } 2628 // compute volume for this track 2629 float left, right; 2630 if (track->isMuted() || mMasterMute || track->isPausing() || 2631 mStreamTypes[track->type()].mute) { 2632 left = right = 0; 2633 if (track->isPausing()) { 2634 track->setPaused(); 2635 } 2636 } else { 2637 float typeVolume = mStreamTypes[track->type()].volume; 2638 float v = mMasterVolume * typeVolume; 2639 float v_clamped = v * cblk->volume[0]; 2640 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2641 left = v_clamped/MAX_GAIN; 2642 v_clamped = v * cblk->volume[1]; 2643 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2644 right = v_clamped/MAX_GAIN; 2645 } 2646 2647 if (left != mLeftVolFloat || right != mRightVolFloat) { 2648 mLeftVolFloat = left; 2649 mRightVolFloat = right; 2650 2651 // If audio HAL implements volume control, 2652 // force software volume to nominal value 2653 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2654 left = 1.0f; 2655 right = 1.0f; 2656 } 2657 2658 // Convert volumes from float to 8.24 2659 uint32_t vl = (uint32_t)(left * (1 << 24)); 2660 uint32_t vr = (uint32_t)(right * (1 << 24)); 2661 2662 // Delegate volume control to effect in track effect chain if needed 2663 // only one effect chain can be present on DirectOutputThread, so if 2664 // there is one, the track is connected to it 2665 if (!effectChains.isEmpty()) { 2666 // Do not ramp volume if volume is controlled by effect 2667 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2668 rampVolume = false; 2669 } 2670 } 2671 2672 // Convert volumes from 8.24 to 4.12 format 2673 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2674 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2675 leftVol = (uint16_t)v_clamped; 2676 v_clamped = (vr + (1 << 11)) >> 12; 2677 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2678 rightVol = (uint16_t)v_clamped; 2679 } else { 2680 leftVol = mLeftVolShort; 2681 rightVol = mRightVolShort; 2682 rampVolume = false; 2683 } 2684 2685 // reset retry count 2686 track->mRetryCount = kMaxTrackRetriesDirect; 2687 activeTrack = t; 2688 mixerStatus = MIXER_TRACKS_READY; 2689 } else { 2690 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2691 if (track->isStopped()) { 2692 track->reset(); 2693 } 2694 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2695 // We have consumed all the buffers of this track. 2696 // Remove it from the list of active tracks. 2697 trackToRemove = track; 2698 } else { 2699 // No buffers for this track. Give it a few chances to 2700 // fill a buffer, then remove it from active list. 2701 if (--(track->mRetryCount) <= 0) { 2702 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2703 trackToRemove = track; 2704 } else { 2705 mixerStatus = MIXER_TRACKS_ENABLED; 2706 } 2707 } 2708 } 2709 } 2710 2711 // remove all the tracks that need to be... 2712 if (UNLIKELY(trackToRemove != 0)) { 2713 mActiveTracks.remove(trackToRemove); 2714 if (!effectChains.isEmpty()) { 2715 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2716 trackToRemove->sessionId()); 2717 effectChains[0]->decActiveTrackCnt(); 2718 } 2719 if (trackToRemove->isTerminated()) { 2720 removeTrack_l(trackToRemove); 2721 } 2722 } 2723 2724 lockEffectChains_l(effectChains); 2725 } 2726 2727 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2728 AudioBufferProvider::Buffer buffer; 2729 size_t frameCount = mFrameCount; 2730 curBuf = (int8_t *)mMixBuffer; 2731 // output audio to hardware 2732 while (frameCount) { 2733 buffer.frameCount = frameCount; 2734 activeTrack->getNextBuffer(&buffer); 2735 if (UNLIKELY(buffer.raw == 0)) { 2736 memset(curBuf, 0, frameCount * mFrameSize); 2737 break; 2738 } 2739 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2740 frameCount -= buffer.frameCount; 2741 curBuf += buffer.frameCount * mFrameSize; 2742 activeTrack->releaseBuffer(&buffer); 2743 } 2744 sleepTime = 0; 2745 standbyTime = systemTime() + standbyDelay; 2746 } else { 2747 if (sleepTime == 0) { 2748 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2749 sleepTime = activeSleepTime; 2750 } else { 2751 sleepTime = idleSleepTime; 2752 } 2753 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2754 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2755 sleepTime = 0; 2756 } 2757 } 2758 2759 if (mSuspended) { 2760 sleepTime = suspendSleepTimeUs(); 2761 } 2762 // sleepTime == 0 means we must write to audio hardware 2763 if (sleepTime == 0) { 2764 if (mixerStatus == MIXER_TRACKS_READY) { 2765 applyVolume(leftVol, rightVol, rampVolume); 2766 } 2767 for (size_t i = 0; i < effectChains.size(); i ++) { 2768 effectChains[i]->process_l(); 2769 } 2770 unlockEffectChains(effectChains); 2771 2772 mLastWriteTime = systemTime(); 2773 mInWrite = true; 2774 mBytesWritten += mixBufferSize; 2775 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2776 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2777 mNumWrites++; 2778 mInWrite = false; 2779 mStandby = false; 2780 } else { 2781 unlockEffectChains(effectChains); 2782 usleep(sleepTime); 2783 } 2784 2785 // finally let go of removed track, without the lock held 2786 // since we can't guarantee the destructors won't acquire that 2787 // same lock. 2788 trackToRemove.clear(); 2789 activeTrack.clear(); 2790 2791 // Effect chains will be actually deleted here if they were removed from 2792 // mEffectChains list during mixing or effects processing 2793 effectChains.clear(); 2794 } 2795 2796 if (!mStandby) { 2797 mOutput->stream->common.standby(&mOutput->stream->common); 2798 } 2799 2800 releaseWakeLock(); 2801 2802 ALOGV("DirectOutputThread %p exiting", this); 2803 return false; 2804} 2805 2806// getTrackName_l() must be called with ThreadBase::mLock held 2807int AudioFlinger::DirectOutputThread::getTrackName_l() 2808{ 2809 return 0; 2810} 2811 2812// deleteTrackName_l() must be called with ThreadBase::mLock held 2813void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2814{ 2815} 2816 2817// checkForNewParameters_l() must be called with ThreadBase::mLock held 2818bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2819{ 2820 bool reconfig = false; 2821 2822 while (!mNewParameters.isEmpty()) { 2823 status_t status = NO_ERROR; 2824 String8 keyValuePair = mNewParameters[0]; 2825 AudioParameter param = AudioParameter(keyValuePair); 2826 int value; 2827 2828 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2829 // do not accept frame count changes if tracks are open as the track buffer 2830 // size depends on frame count and correct behavior would not be garantied 2831 // if frame count is changed after track creation 2832 if (!mTracks.isEmpty()) { 2833 status = INVALID_OPERATION; 2834 } else { 2835 reconfig = true; 2836 } 2837 } 2838 if (status == NO_ERROR) { 2839 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2840 keyValuePair.string()); 2841 if (!mStandby && status == INVALID_OPERATION) { 2842 mOutput->stream->common.standby(&mOutput->stream->common); 2843 mStandby = true; 2844 mBytesWritten = 0; 2845 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2846 keyValuePair.string()); 2847 } 2848 if (status == NO_ERROR && reconfig) { 2849 readOutputParameters(); 2850 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2851 } 2852 } 2853 2854 mNewParameters.removeAt(0); 2855 2856 mParamStatus = status; 2857 mParamCond.signal(); 2858 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2859 // already timed out waiting for the status and will never signal the condition. 2860 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 2861 } 2862 return reconfig; 2863} 2864 2865uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2866{ 2867 uint32_t time; 2868 if (audio_is_linear_pcm(mFormat)) { 2869 time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2870 } else { 2871 time = 10000; 2872 } 2873 return time; 2874} 2875 2876uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2877{ 2878 uint32_t time; 2879 if (audio_is_linear_pcm(mFormat)) { 2880 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2881 } else { 2882 time = 10000; 2883 } 2884 return time; 2885} 2886 2887uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2888{ 2889 uint32_t time; 2890 if (audio_is_linear_pcm(mFormat)) { 2891 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2892 } else { 2893 time = 10000; 2894 } 2895 return time; 2896} 2897 2898 2899// ---------------------------------------------------------------------------- 2900 2901AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2902 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2903{ 2904 mType = ThreadBase::DUPLICATING; 2905 addOutputTrack(mainThread); 2906} 2907 2908AudioFlinger::DuplicatingThread::~DuplicatingThread() 2909{ 2910 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2911 mOutputTracks[i]->destroy(); 2912 } 2913 mOutputTracks.clear(); 2914} 2915 2916bool AudioFlinger::DuplicatingThread::threadLoop() 2917{ 2918 Vector< sp<Track> > tracksToRemove; 2919 uint32_t mixerStatus = MIXER_IDLE; 2920 nsecs_t standbyTime = systemTime(); 2921 size_t mixBufferSize = mFrameCount*mFrameSize; 2922 SortedVector< sp<OutputTrack> > outputTracks; 2923 uint32_t writeFrames = 0; 2924 uint32_t activeSleepTime = activeSleepTimeUs(); 2925 uint32_t idleSleepTime = idleSleepTimeUs(); 2926 uint32_t sleepTime = idleSleepTime; 2927 Vector< sp<EffectChain> > effectChains; 2928 2929 acquireWakeLock(); 2930 2931 while (!exitPending()) 2932 { 2933 processConfigEvents(); 2934 2935 mixerStatus = MIXER_IDLE; 2936 { // scope for the mLock 2937 2938 Mutex::Autolock _l(mLock); 2939 2940 if (checkForNewParameters_l()) { 2941 mixBufferSize = mFrameCount*mFrameSize; 2942 updateWaitTime(); 2943 activeSleepTime = activeSleepTimeUs(); 2944 idleSleepTime = idleSleepTimeUs(); 2945 } 2946 2947 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2948 2949 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2950 outputTracks.add(mOutputTracks[i]); 2951 } 2952 2953 // put audio hardware into standby after short delay 2954 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2955 mSuspended) { 2956 if (!mStandby) { 2957 for (size_t i = 0; i < outputTracks.size(); i++) { 2958 outputTracks[i]->stop(); 2959 } 2960 mStandby = true; 2961 mBytesWritten = 0; 2962 } 2963 2964 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2965 // we're about to wait, flush the binder command buffer 2966 IPCThreadState::self()->flushCommands(); 2967 outputTracks.clear(); 2968 2969 if (exitPending()) break; 2970 2971 releaseWakeLock_l(); 2972 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2973 mWaitWorkCV.wait(mLock); 2974 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2975 acquireWakeLock_l(); 2976 2977 if (mMasterMute == false) { 2978 char value[PROPERTY_VALUE_MAX]; 2979 property_get("ro.audio.silent", value, "0"); 2980 if (atoi(value)) { 2981 LOGD("Silence is golden"); 2982 setMasterMute(true); 2983 } 2984 } 2985 2986 standbyTime = systemTime() + kStandbyTimeInNsecs; 2987 sleepTime = idleSleepTime; 2988 continue; 2989 } 2990 } 2991 2992 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2993 2994 // prevent any changes in effect chain list and in each effect chain 2995 // during mixing and effect process as the audio buffers could be deleted 2996 // or modified if an effect is created or deleted 2997 lockEffectChains_l(effectChains); 2998 } 2999 3000 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3001 // mix buffers... 3002 if (outputsReady(outputTracks)) { 3003 mAudioMixer->process(); 3004 } else { 3005 memset(mMixBuffer, 0, mixBufferSize); 3006 } 3007 sleepTime = 0; 3008 writeFrames = mFrameCount; 3009 } else { 3010 if (sleepTime == 0) { 3011 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3012 sleepTime = activeSleepTime; 3013 } else { 3014 sleepTime = idleSleepTime; 3015 } 3016 } else if (mBytesWritten != 0) { 3017 // flush remaining overflow buffers in output tracks 3018 for (size_t i = 0; i < outputTracks.size(); i++) { 3019 if (outputTracks[i]->isActive()) { 3020 sleepTime = 0; 3021 writeFrames = 0; 3022 memset(mMixBuffer, 0, mixBufferSize); 3023 break; 3024 } 3025 } 3026 } 3027 } 3028 3029 if (mSuspended) { 3030 sleepTime = suspendSleepTimeUs(); 3031 } 3032 // sleepTime == 0 means we must write to audio hardware 3033 if (sleepTime == 0) { 3034 for (size_t i = 0; i < effectChains.size(); i ++) { 3035 effectChains[i]->process_l(); 3036 } 3037 // enable changes in effect chain 3038 unlockEffectChains(effectChains); 3039 3040 standbyTime = systemTime() + kStandbyTimeInNsecs; 3041 for (size_t i = 0; i < outputTracks.size(); i++) { 3042 outputTracks[i]->write(mMixBuffer, writeFrames); 3043 } 3044 mStandby = false; 3045 mBytesWritten += mixBufferSize; 3046 } else { 3047 // enable changes in effect chain 3048 unlockEffectChains(effectChains); 3049 usleep(sleepTime); 3050 } 3051 3052 // finally let go of all our tracks, without the lock held 3053 // since we can't guarantee the destructors won't acquire that 3054 // same lock. 3055 tracksToRemove.clear(); 3056 outputTracks.clear(); 3057 3058 // Effect chains will be actually deleted here if they were removed from 3059 // mEffectChains list during mixing or effects processing 3060 effectChains.clear(); 3061 } 3062 3063 releaseWakeLock(); 3064 3065 return false; 3066} 3067 3068void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3069{ 3070 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3071 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3072 this, 3073 mSampleRate, 3074 mFormat, 3075 mChannelMask, 3076 frameCount); 3077 if (outputTrack->cblk() != NULL) { 3078 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3079 mOutputTracks.add(outputTrack); 3080 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3081 updateWaitTime(); 3082 } 3083} 3084 3085void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3086{ 3087 Mutex::Autolock _l(mLock); 3088 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3089 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3090 mOutputTracks[i]->destroy(); 3091 mOutputTracks.removeAt(i); 3092 updateWaitTime(); 3093 return; 3094 } 3095 } 3096 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3097} 3098 3099void AudioFlinger::DuplicatingThread::updateWaitTime() 3100{ 3101 mWaitTimeMs = UINT_MAX; 3102 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3103 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3104 if (strong != NULL) { 3105 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3106 if (waitTimeMs < mWaitTimeMs) { 3107 mWaitTimeMs = waitTimeMs; 3108 } 3109 } 3110 } 3111} 3112 3113 3114bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3115{ 3116 for (size_t i = 0; i < outputTracks.size(); i++) { 3117 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3118 if (thread == 0) { 3119 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3120 return false; 3121 } 3122 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3123 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3124 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3125 return false; 3126 } 3127 } 3128 return true; 3129} 3130 3131uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3132{ 3133 return (mWaitTimeMs * 1000) / 2; 3134} 3135 3136// ---------------------------------------------------------------------------- 3137 3138// TrackBase constructor must be called with AudioFlinger::mLock held 3139AudioFlinger::ThreadBase::TrackBase::TrackBase( 3140 const wp<ThreadBase>& thread, 3141 const sp<Client>& client, 3142 uint32_t sampleRate, 3143 uint32_t format, 3144 uint32_t channelMask, 3145 int frameCount, 3146 uint32_t flags, 3147 const sp<IMemory>& sharedBuffer, 3148 int sessionId) 3149 : RefBase(), 3150 mThread(thread), 3151 mClient(client), 3152 mCblk(0), 3153 mFrameCount(0), 3154 mState(IDLE), 3155 mClientTid(-1), 3156 mFormat(format), 3157 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3158 mSessionId(sessionId) 3159{ 3160 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3161 3162 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3163 size_t size = sizeof(audio_track_cblk_t); 3164 uint8_t channelCount = popcount(channelMask); 3165 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3166 if (sharedBuffer == 0) { 3167 size += bufferSize; 3168 } 3169 3170 if (client != NULL) { 3171 mCblkMemory = client->heap()->allocate(size); 3172 if (mCblkMemory != 0) { 3173 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3174 if (mCblk) { // construct the shared structure in-place. 3175 new(mCblk) audio_track_cblk_t(); 3176 // clear all buffers 3177 mCblk->frameCount = frameCount; 3178 mCblk->sampleRate = sampleRate; 3179 mChannelCount = channelCount; 3180 mChannelMask = channelMask; 3181 if (sharedBuffer == 0) { 3182 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3183 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3184 // Force underrun condition to avoid false underrun callback until first data is 3185 // written to buffer (other flags are cleared) 3186 mCblk->flags = CBLK_UNDERRUN_ON; 3187 } else { 3188 mBuffer = sharedBuffer->pointer(); 3189 } 3190 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3191 } 3192 } else { 3193 LOGE("not enough memory for AudioTrack size=%u", size); 3194 client->heap()->dump("AudioTrack"); 3195 return; 3196 } 3197 } else { 3198 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3199 if (mCblk) { // construct the shared structure in-place. 3200 new(mCblk) audio_track_cblk_t(); 3201 // clear all buffers 3202 mCblk->frameCount = frameCount; 3203 mCblk->sampleRate = sampleRate; 3204 mChannelCount = channelCount; 3205 mChannelMask = channelMask; 3206 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3207 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3208 // Force underrun condition to avoid false underrun callback until first data is 3209 // written to buffer (other flags are cleared) 3210 mCblk->flags = CBLK_UNDERRUN_ON; 3211 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3212 } 3213 } 3214} 3215 3216AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3217{ 3218 if (mCblk) { 3219 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3220 if (mClient == NULL) { 3221 delete mCblk; 3222 } 3223 } 3224 mCblkMemory.clear(); // and free the shared memory 3225 if (mClient != NULL) { 3226 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3227 mClient.clear(); 3228 } 3229} 3230 3231void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3232{ 3233 buffer->raw = 0; 3234 mFrameCount = buffer->frameCount; 3235 step(); 3236 buffer->frameCount = 0; 3237} 3238 3239bool AudioFlinger::ThreadBase::TrackBase::step() { 3240 bool result; 3241 audio_track_cblk_t* cblk = this->cblk(); 3242 3243 result = cblk->stepServer(mFrameCount); 3244 if (!result) { 3245 ALOGV("stepServer failed acquiring cblk mutex"); 3246 mFlags |= STEPSERVER_FAILED; 3247 } 3248 return result; 3249} 3250 3251void AudioFlinger::ThreadBase::TrackBase::reset() { 3252 audio_track_cblk_t* cblk = this->cblk(); 3253 3254 cblk->user = 0; 3255 cblk->server = 0; 3256 cblk->userBase = 0; 3257 cblk->serverBase = 0; 3258 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3259 ALOGV("TrackBase::reset"); 3260} 3261 3262sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3263{ 3264 return mCblkMemory; 3265} 3266 3267int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3268 return (int)mCblk->sampleRate; 3269} 3270 3271int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3272 return (const int)mChannelCount; 3273} 3274 3275uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3276 return mChannelMask; 3277} 3278 3279void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3280 audio_track_cblk_t* cblk = this->cblk(); 3281 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 3282 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 3283 3284 // Check validity of returned pointer in case the track control block would have been corrupted. 3285 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3286 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 3287 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3288 server %d, serverBase %d, user %d, userBase %d", 3289 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3290 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3291 return 0; 3292 } 3293 3294 return bufferStart; 3295} 3296 3297// ---------------------------------------------------------------------------- 3298 3299// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3300AudioFlinger::PlaybackThread::Track::Track( 3301 const wp<ThreadBase>& thread, 3302 const sp<Client>& client, 3303 int streamType, 3304 uint32_t sampleRate, 3305 uint32_t format, 3306 uint32_t channelMask, 3307 int frameCount, 3308 const sp<IMemory>& sharedBuffer, 3309 int sessionId) 3310 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3311 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3312 mAuxEffectId(0), mHasVolumeController(false) 3313{ 3314 if (mCblk != NULL) { 3315 sp<ThreadBase> baseThread = thread.promote(); 3316 if (baseThread != 0) { 3317 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3318 mName = playbackThread->getTrackName_l(); 3319 mMainBuffer = playbackThread->mixBuffer(); 3320 } 3321 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3322 if (mName < 0) { 3323 LOGE("no more track names available"); 3324 } 3325 mVolume[0] = 1.0f; 3326 mVolume[1] = 1.0f; 3327 mStreamType = streamType; 3328 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3329 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3330 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3331 } 3332} 3333 3334AudioFlinger::PlaybackThread::Track::~Track() 3335{ 3336 ALOGV("PlaybackThread::Track destructor"); 3337 sp<ThreadBase> thread = mThread.promote(); 3338 if (thread != 0) { 3339 Mutex::Autolock _l(thread->mLock); 3340 mState = TERMINATED; 3341 } 3342} 3343 3344void AudioFlinger::PlaybackThread::Track::destroy() 3345{ 3346 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3347 // by removing it from mTracks vector, so there is a risk that this Tracks's 3348 // desctructor is called. As the destructor needs to lock mLock, 3349 // we must acquire a strong reference on this Track before locking mLock 3350 // here so that the destructor is called only when exiting this function. 3351 // On the other hand, as long as Track::destroy() is only called by 3352 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3353 // this Track with its member mTrack. 3354 sp<Track> keep(this); 3355 { // scope for mLock 3356 sp<ThreadBase> thread = mThread.promote(); 3357 if (thread != 0) { 3358 if (!isOutputTrack()) { 3359 if (mState == ACTIVE || mState == RESUMING) { 3360 AudioSystem::stopOutput(thread->id(), 3361 (audio_stream_type_t)mStreamType, 3362 mSessionId); 3363 3364 // to track the speaker usage 3365 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3366 } 3367 AudioSystem::releaseOutput(thread->id()); 3368 } 3369 Mutex::Autolock _l(thread->mLock); 3370 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3371 playbackThread->destroyTrack_l(this); 3372 } 3373 } 3374} 3375 3376void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3377{ 3378 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", 3379 mName - AudioMixer::TRACK0, 3380 (mClient == NULL) ? getpid() : mClient->pid(), 3381 mStreamType, 3382 mFormat, 3383 mChannelMask, 3384 mSessionId, 3385 mFrameCount, 3386 mState, 3387 mMute, 3388 mFillingUpStatus, 3389 mCblk->sampleRate, 3390 mCblk->volume[0], 3391 mCblk->volume[1], 3392 mCblk->server, 3393 mCblk->user, 3394 (int)mMainBuffer, 3395 (int)mAuxBuffer); 3396} 3397 3398status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3399{ 3400 audio_track_cblk_t* cblk = this->cblk(); 3401 uint32_t framesReady; 3402 uint32_t framesReq = buffer->frameCount; 3403 3404 // Check if last stepServer failed, try to step now 3405 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3406 if (!step()) goto getNextBuffer_exit; 3407 ALOGV("stepServer recovered"); 3408 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3409 } 3410 3411 framesReady = cblk->framesReady(); 3412 3413 if (LIKELY(framesReady)) { 3414 uint32_t s = cblk->server; 3415 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3416 3417 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3418 if (framesReq > framesReady) { 3419 framesReq = framesReady; 3420 } 3421 if (s + framesReq > bufferEnd) { 3422 framesReq = bufferEnd - s; 3423 } 3424 3425 buffer->raw = getBuffer(s, framesReq); 3426 if (buffer->raw == 0) goto getNextBuffer_exit; 3427 3428 buffer->frameCount = framesReq; 3429 return NO_ERROR; 3430 } 3431 3432getNextBuffer_exit: 3433 buffer->raw = 0; 3434 buffer->frameCount = 0; 3435 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3436 return NOT_ENOUGH_DATA; 3437} 3438 3439bool AudioFlinger::PlaybackThread::Track::isReady() const { 3440 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3441 3442 if (mCblk->framesReady() >= mCblk->frameCount || 3443 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3444 mFillingUpStatus = FS_FILLED; 3445 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3446 return true; 3447 } 3448 return false; 3449} 3450 3451status_t AudioFlinger::PlaybackThread::Track::start() 3452{ 3453 status_t status = NO_ERROR; 3454 ALOGV("start(%d), calling thread %d session %d", 3455 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3456 sp<ThreadBase> thread = mThread.promote(); 3457 if (thread != 0) { 3458 Mutex::Autolock _l(thread->mLock); 3459 int state = mState; 3460 // here the track could be either new, or restarted 3461 // in both cases "unstop" the track 3462 if (mState == PAUSED) { 3463 mState = TrackBase::RESUMING; 3464 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3465 } else { 3466 mState = TrackBase::ACTIVE; 3467 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3468 } 3469 3470 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3471 thread->mLock.unlock(); 3472 status = AudioSystem::startOutput(thread->id(), 3473 (audio_stream_type_t)mStreamType, 3474 mSessionId); 3475 thread->mLock.lock(); 3476 3477 // to track the speaker usage 3478 if (status == NO_ERROR) { 3479 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3480 } 3481 } 3482 if (status == NO_ERROR) { 3483 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3484 playbackThread->addTrack_l(this); 3485 } else { 3486 mState = state; 3487 } 3488 } else { 3489 status = BAD_VALUE; 3490 } 3491 return status; 3492} 3493 3494void AudioFlinger::PlaybackThread::Track::stop() 3495{ 3496 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3497 sp<ThreadBase> thread = mThread.promote(); 3498 if (thread != 0) { 3499 Mutex::Autolock _l(thread->mLock); 3500 int state = mState; 3501 if (mState > STOPPED) { 3502 mState = STOPPED; 3503 // If the track is not active (PAUSED and buffers full), flush buffers 3504 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3505 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3506 reset(); 3507 } 3508 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3509 } 3510 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3511 thread->mLock.unlock(); 3512 AudioSystem::stopOutput(thread->id(), 3513 (audio_stream_type_t)mStreamType, 3514 mSessionId); 3515 thread->mLock.lock(); 3516 3517 // to track the speaker usage 3518 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3519 } 3520 } 3521} 3522 3523void AudioFlinger::PlaybackThread::Track::pause() 3524{ 3525 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3526 sp<ThreadBase> thread = mThread.promote(); 3527 if (thread != 0) { 3528 Mutex::Autolock _l(thread->mLock); 3529 if (mState == ACTIVE || mState == RESUMING) { 3530 mState = PAUSING; 3531 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3532 if (!isOutputTrack()) { 3533 thread->mLock.unlock(); 3534 AudioSystem::stopOutput(thread->id(), 3535 (audio_stream_type_t)mStreamType, 3536 mSessionId); 3537 thread->mLock.lock(); 3538 3539 // to track the speaker usage 3540 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3541 } 3542 } 3543 } 3544} 3545 3546void AudioFlinger::PlaybackThread::Track::flush() 3547{ 3548 ALOGV("flush(%d)", mName); 3549 sp<ThreadBase> thread = mThread.promote(); 3550 if (thread != 0) { 3551 Mutex::Autolock _l(thread->mLock); 3552 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3553 return; 3554 } 3555 // No point remaining in PAUSED state after a flush => go to 3556 // STOPPED state 3557 mState = STOPPED; 3558 3559 // do not reset the track if it is still in the process of being stopped or paused. 3560 // this will be done by prepareTracks_l() when the track is stopped. 3561 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3562 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3563 reset(); 3564 } 3565 } 3566} 3567 3568void AudioFlinger::PlaybackThread::Track::reset() 3569{ 3570 // Do not reset twice to avoid discarding data written just after a flush and before 3571 // the audioflinger thread detects the track is stopped. 3572 if (!mResetDone) { 3573 TrackBase::reset(); 3574 // Force underrun condition to avoid false underrun callback until first data is 3575 // written to buffer 3576 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3577 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3578 mFillingUpStatus = FS_FILLING; 3579 mResetDone = true; 3580 } 3581} 3582 3583void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3584{ 3585 mMute = muted; 3586} 3587 3588void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3589{ 3590 mVolume[0] = left; 3591 mVolume[1] = right; 3592} 3593 3594status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3595{ 3596 status_t status = DEAD_OBJECT; 3597 sp<ThreadBase> thread = mThread.promote(); 3598 if (thread != 0) { 3599 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3600 status = playbackThread->attachAuxEffect(this, EffectId); 3601 } 3602 return status; 3603} 3604 3605void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3606{ 3607 mAuxEffectId = EffectId; 3608 mAuxBuffer = buffer; 3609} 3610 3611// ---------------------------------------------------------------------------- 3612 3613// RecordTrack constructor must be called with AudioFlinger::mLock held 3614AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3615 const wp<ThreadBase>& thread, 3616 const sp<Client>& client, 3617 uint32_t sampleRate, 3618 uint32_t format, 3619 uint32_t channelMask, 3620 int frameCount, 3621 uint32_t flags, 3622 int sessionId) 3623 : TrackBase(thread, client, sampleRate, format, 3624 channelMask, frameCount, flags, 0, sessionId), 3625 mOverflow(false) 3626{ 3627 if (mCblk != NULL) { 3628 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3629 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3630 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3631 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3632 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3633 } else { 3634 mCblk->frameSize = sizeof(int8_t); 3635 } 3636 } 3637} 3638 3639AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3640{ 3641 sp<ThreadBase> thread = mThread.promote(); 3642 if (thread != 0) { 3643 AudioSystem::releaseInput(thread->id()); 3644 } 3645} 3646 3647status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3648{ 3649 audio_track_cblk_t* cblk = this->cblk(); 3650 uint32_t framesAvail; 3651 uint32_t framesReq = buffer->frameCount; 3652 3653 // Check if last stepServer failed, try to step now 3654 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3655 if (!step()) goto getNextBuffer_exit; 3656 ALOGV("stepServer recovered"); 3657 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3658 } 3659 3660 framesAvail = cblk->framesAvailable_l(); 3661 3662 if (LIKELY(framesAvail)) { 3663 uint32_t s = cblk->server; 3664 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3665 3666 if (framesReq > framesAvail) { 3667 framesReq = framesAvail; 3668 } 3669 if (s + framesReq > bufferEnd) { 3670 framesReq = bufferEnd - s; 3671 } 3672 3673 buffer->raw = getBuffer(s, framesReq); 3674 if (buffer->raw == 0) goto getNextBuffer_exit; 3675 3676 buffer->frameCount = framesReq; 3677 return NO_ERROR; 3678 } 3679 3680getNextBuffer_exit: 3681 buffer->raw = 0; 3682 buffer->frameCount = 0; 3683 return NOT_ENOUGH_DATA; 3684} 3685 3686status_t AudioFlinger::RecordThread::RecordTrack::start() 3687{ 3688 sp<ThreadBase> thread = mThread.promote(); 3689 if (thread != 0) { 3690 RecordThread *recordThread = (RecordThread *)thread.get(); 3691 return recordThread->start(this); 3692 } else { 3693 return BAD_VALUE; 3694 } 3695} 3696 3697void AudioFlinger::RecordThread::RecordTrack::stop() 3698{ 3699 sp<ThreadBase> thread = mThread.promote(); 3700 if (thread != 0) { 3701 RecordThread *recordThread = (RecordThread *)thread.get(); 3702 recordThread->stop(this); 3703 TrackBase::reset(); 3704 // Force overerrun condition to avoid false overrun callback until first data is 3705 // read from buffer 3706 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3707 } 3708} 3709 3710void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3711{ 3712 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3713 (mClient == NULL) ? getpid() : mClient->pid(), 3714 mFormat, 3715 mChannelMask, 3716 mSessionId, 3717 mFrameCount, 3718 mState, 3719 mCblk->sampleRate, 3720 mCblk->server, 3721 mCblk->user); 3722} 3723 3724 3725// ---------------------------------------------------------------------------- 3726 3727AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3728 const wp<ThreadBase>& thread, 3729 DuplicatingThread *sourceThread, 3730 uint32_t sampleRate, 3731 uint32_t format, 3732 uint32_t channelMask, 3733 int frameCount) 3734 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3735 mActive(false), mSourceThread(sourceThread) 3736{ 3737 3738 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3739 if (mCblk != NULL) { 3740 mCblk->flags |= CBLK_DIRECTION_OUT; 3741 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3742 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3743 mOutBuffer.frameCount = 0; 3744 playbackThread->mTracks.add(this); 3745 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3746 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3747 mCblk, mBuffer, mCblk->buffers, 3748 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3749 } else { 3750 LOGW("Error creating output track on thread %p", playbackThread); 3751 } 3752} 3753 3754AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3755{ 3756 clearBufferQueue(); 3757} 3758 3759status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3760{ 3761 status_t status = Track::start(); 3762 if (status != NO_ERROR) { 3763 return status; 3764 } 3765 3766 mActive = true; 3767 mRetryCount = 127; 3768 return status; 3769} 3770 3771void AudioFlinger::PlaybackThread::OutputTrack::stop() 3772{ 3773 Track::stop(); 3774 clearBufferQueue(); 3775 mOutBuffer.frameCount = 0; 3776 mActive = false; 3777} 3778 3779bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3780{ 3781 Buffer *pInBuffer; 3782 Buffer inBuffer; 3783 uint32_t channelCount = mChannelCount; 3784 bool outputBufferFull = false; 3785 inBuffer.frameCount = frames; 3786 inBuffer.i16 = data; 3787 3788 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3789 3790 if (!mActive && frames != 0) { 3791 start(); 3792 sp<ThreadBase> thread = mThread.promote(); 3793 if (thread != 0) { 3794 MixerThread *mixerThread = (MixerThread *)thread.get(); 3795 if (mCblk->frameCount > frames){ 3796 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3797 uint32_t startFrames = (mCblk->frameCount - frames); 3798 pInBuffer = new Buffer; 3799 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3800 pInBuffer->frameCount = startFrames; 3801 pInBuffer->i16 = pInBuffer->mBuffer; 3802 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3803 mBufferQueue.add(pInBuffer); 3804 } else { 3805 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3806 } 3807 } 3808 } 3809 } 3810 3811 while (waitTimeLeftMs) { 3812 // First write pending buffers, then new data 3813 if (mBufferQueue.size()) { 3814 pInBuffer = mBufferQueue.itemAt(0); 3815 } else { 3816 pInBuffer = &inBuffer; 3817 } 3818 3819 if (pInBuffer->frameCount == 0) { 3820 break; 3821 } 3822 3823 if (mOutBuffer.frameCount == 0) { 3824 mOutBuffer.frameCount = pInBuffer->frameCount; 3825 nsecs_t startTime = systemTime(); 3826 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3827 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3828 outputBufferFull = true; 3829 break; 3830 } 3831 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3832 if (waitTimeLeftMs >= waitTimeMs) { 3833 waitTimeLeftMs -= waitTimeMs; 3834 } else { 3835 waitTimeLeftMs = 0; 3836 } 3837 } 3838 3839 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3840 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3841 mCblk->stepUser(outFrames); 3842 pInBuffer->frameCount -= outFrames; 3843 pInBuffer->i16 += outFrames * channelCount; 3844 mOutBuffer.frameCount -= outFrames; 3845 mOutBuffer.i16 += outFrames * channelCount; 3846 3847 if (pInBuffer->frameCount == 0) { 3848 if (mBufferQueue.size()) { 3849 mBufferQueue.removeAt(0); 3850 delete [] pInBuffer->mBuffer; 3851 delete pInBuffer; 3852 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3853 } else { 3854 break; 3855 } 3856 } 3857 } 3858 3859 // If we could not write all frames, allocate a buffer and queue it for next time. 3860 if (inBuffer.frameCount) { 3861 sp<ThreadBase> thread = mThread.promote(); 3862 if (thread != 0 && !thread->standby()) { 3863 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3864 pInBuffer = new Buffer; 3865 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3866 pInBuffer->frameCount = inBuffer.frameCount; 3867 pInBuffer->i16 = pInBuffer->mBuffer; 3868 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3869 mBufferQueue.add(pInBuffer); 3870 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3871 } else { 3872 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3873 } 3874 } 3875 } 3876 3877 // Calling write() with a 0 length buffer, means that no more data will be written: 3878 // If no more buffers are pending, fill output track buffer to make sure it is started 3879 // by output mixer. 3880 if (frames == 0 && mBufferQueue.size() == 0) { 3881 if (mCblk->user < mCblk->frameCount) { 3882 frames = mCblk->frameCount - mCblk->user; 3883 pInBuffer = new Buffer; 3884 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3885 pInBuffer->frameCount = frames; 3886 pInBuffer->i16 = pInBuffer->mBuffer; 3887 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3888 mBufferQueue.add(pInBuffer); 3889 } else if (mActive) { 3890 stop(); 3891 } 3892 } 3893 3894 return outputBufferFull; 3895} 3896 3897status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3898{ 3899 int active; 3900 status_t result; 3901 audio_track_cblk_t* cblk = mCblk; 3902 uint32_t framesReq = buffer->frameCount; 3903 3904// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3905 buffer->frameCount = 0; 3906 3907 uint32_t framesAvail = cblk->framesAvailable(); 3908 3909 3910 if (framesAvail == 0) { 3911 Mutex::Autolock _l(cblk->lock); 3912 goto start_loop_here; 3913 while (framesAvail == 0) { 3914 active = mActive; 3915 if (UNLIKELY(!active)) { 3916 ALOGV("Not active and NO_MORE_BUFFERS"); 3917 return AudioTrack::NO_MORE_BUFFERS; 3918 } 3919 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3920 if (result != NO_ERROR) { 3921 return AudioTrack::NO_MORE_BUFFERS; 3922 } 3923 // read the server count again 3924 start_loop_here: 3925 framesAvail = cblk->framesAvailable_l(); 3926 } 3927 } 3928 3929// if (framesAvail < framesReq) { 3930// return AudioTrack::NO_MORE_BUFFERS; 3931// } 3932 3933 if (framesReq > framesAvail) { 3934 framesReq = framesAvail; 3935 } 3936 3937 uint32_t u = cblk->user; 3938 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3939 3940 if (u + framesReq > bufferEnd) { 3941 framesReq = bufferEnd - u; 3942 } 3943 3944 buffer->frameCount = framesReq; 3945 buffer->raw = (void *)cblk->buffer(u); 3946 return NO_ERROR; 3947} 3948 3949 3950void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3951{ 3952 size_t size = mBufferQueue.size(); 3953 Buffer *pBuffer; 3954 3955 for (size_t i = 0; i < size; i++) { 3956 pBuffer = mBufferQueue.itemAt(i); 3957 delete [] pBuffer->mBuffer; 3958 delete pBuffer; 3959 } 3960 mBufferQueue.clear(); 3961} 3962 3963// ---------------------------------------------------------------------------- 3964 3965AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3966 : RefBase(), 3967 mAudioFlinger(audioFlinger), 3968 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3969 mPid(pid) 3970{ 3971 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3972} 3973 3974// Client destructor must be called with AudioFlinger::mLock held 3975AudioFlinger::Client::~Client() 3976{ 3977 mAudioFlinger->removeClient_l(mPid); 3978} 3979 3980const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3981{ 3982 return mMemoryDealer; 3983} 3984 3985// ---------------------------------------------------------------------------- 3986 3987AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3988 const sp<IAudioFlingerClient>& client, 3989 pid_t pid) 3990 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3991{ 3992} 3993 3994AudioFlinger::NotificationClient::~NotificationClient() 3995{ 3996 mClient.clear(); 3997} 3998 3999void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4000{ 4001 sp<NotificationClient> keep(this); 4002 { 4003 mAudioFlinger->removeNotificationClient(mPid); 4004 } 4005} 4006 4007// ---------------------------------------------------------------------------- 4008 4009AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4010 : BnAudioTrack(), 4011 mTrack(track) 4012{ 4013} 4014 4015AudioFlinger::TrackHandle::~TrackHandle() { 4016 // just stop the track on deletion, associated resources 4017 // will be freed from the main thread once all pending buffers have 4018 // been played. Unless it's not in the active track list, in which 4019 // case we free everything now... 4020 mTrack->destroy(); 4021} 4022 4023status_t AudioFlinger::TrackHandle::start() { 4024 return mTrack->start(); 4025} 4026 4027void AudioFlinger::TrackHandle::stop() { 4028 mTrack->stop(); 4029} 4030 4031void AudioFlinger::TrackHandle::flush() { 4032 mTrack->flush(); 4033} 4034 4035void AudioFlinger::TrackHandle::mute(bool e) { 4036 mTrack->mute(e); 4037} 4038 4039void AudioFlinger::TrackHandle::pause() { 4040 mTrack->pause(); 4041} 4042 4043void AudioFlinger::TrackHandle::setVolume(float left, float right) { 4044 mTrack->setVolume(left, right); 4045} 4046 4047sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4048 return mTrack->getCblk(); 4049} 4050 4051status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4052{ 4053 return mTrack->attachAuxEffect(EffectId); 4054} 4055 4056status_t AudioFlinger::TrackHandle::onTransact( 4057 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4058{ 4059 return BnAudioTrack::onTransact(code, data, reply, flags); 4060} 4061 4062// ---------------------------------------------------------------------------- 4063 4064sp<IAudioRecord> AudioFlinger::openRecord( 4065 pid_t pid, 4066 int input, 4067 uint32_t sampleRate, 4068 uint32_t format, 4069 uint32_t channelMask, 4070 int frameCount, 4071 uint32_t flags, 4072 int *sessionId, 4073 status_t *status) 4074{ 4075 sp<RecordThread::RecordTrack> recordTrack; 4076 sp<RecordHandle> recordHandle; 4077 sp<Client> client; 4078 wp<Client> wclient; 4079 status_t lStatus; 4080 RecordThread *thread; 4081 size_t inFrameCount; 4082 int lSessionId; 4083 4084 // check calling permissions 4085 if (!recordingAllowed()) { 4086 lStatus = PERMISSION_DENIED; 4087 goto Exit; 4088 } 4089 4090 // add client to list 4091 { // scope for mLock 4092 Mutex::Autolock _l(mLock); 4093 thread = checkRecordThread_l(input); 4094 if (thread == NULL) { 4095 lStatus = BAD_VALUE; 4096 goto Exit; 4097 } 4098 4099 wclient = mClients.valueFor(pid); 4100 if (wclient != NULL) { 4101 client = wclient.promote(); 4102 } else { 4103 client = new Client(this, pid); 4104 mClients.add(pid, client); 4105 } 4106 4107 // If no audio session id is provided, create one here 4108 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4109 lSessionId = *sessionId; 4110 } else { 4111 lSessionId = nextUniqueId(); 4112 if (sessionId != NULL) { 4113 *sessionId = lSessionId; 4114 } 4115 } 4116 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4117 recordTrack = thread->createRecordTrack_l(client, 4118 sampleRate, 4119 format, 4120 channelMask, 4121 frameCount, 4122 flags, 4123 lSessionId, 4124 &lStatus); 4125 } 4126 if (lStatus != NO_ERROR) { 4127 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4128 // destructor is called by the TrackBase destructor with mLock held 4129 client.clear(); 4130 recordTrack.clear(); 4131 goto Exit; 4132 } 4133 4134 // return to handle to client 4135 recordHandle = new RecordHandle(recordTrack); 4136 lStatus = NO_ERROR; 4137 4138Exit: 4139 if (status) { 4140 *status = lStatus; 4141 } 4142 return recordHandle; 4143} 4144 4145// ---------------------------------------------------------------------------- 4146 4147AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4148 : BnAudioRecord(), 4149 mRecordTrack(recordTrack) 4150{ 4151} 4152 4153AudioFlinger::RecordHandle::~RecordHandle() { 4154 stop(); 4155} 4156 4157status_t AudioFlinger::RecordHandle::start() { 4158 ALOGV("RecordHandle::start()"); 4159 return mRecordTrack->start(); 4160} 4161 4162void AudioFlinger::RecordHandle::stop() { 4163 ALOGV("RecordHandle::stop()"); 4164 mRecordTrack->stop(); 4165} 4166 4167sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4168 return mRecordTrack->getCblk(); 4169} 4170 4171status_t AudioFlinger::RecordHandle::onTransact( 4172 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4173{ 4174 return BnAudioRecord::onTransact(code, data, reply, flags); 4175} 4176 4177// ---------------------------------------------------------------------------- 4178 4179AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4180 AudioStreamIn *input, 4181 uint32_t sampleRate, 4182 uint32_t channels, 4183 int id, 4184 uint32_t device) : 4185 ThreadBase(audioFlinger, id, device), 4186 mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 4187{ 4188 mType = ThreadBase::RECORD; 4189 4190 snprintf(mName, kNameLength, "AudioIn_%d", id); 4191 4192 mReqChannelCount = popcount(channels); 4193 mReqSampleRate = sampleRate; 4194 readInputParameters(); 4195} 4196 4197 4198AudioFlinger::RecordThread::~RecordThread() 4199{ 4200 delete[] mRsmpInBuffer; 4201 if (mResampler != 0) { 4202 delete mResampler; 4203 delete[] mRsmpOutBuffer; 4204 } 4205} 4206 4207void AudioFlinger::RecordThread::onFirstRef() 4208{ 4209 run(mName, PRIORITY_URGENT_AUDIO); 4210} 4211 4212status_t AudioFlinger::RecordThread::readyToRun() 4213{ 4214 status_t status = initCheck(); 4215 LOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4216 return status; 4217} 4218 4219bool AudioFlinger::RecordThread::threadLoop() 4220{ 4221 AudioBufferProvider::Buffer buffer; 4222 sp<RecordTrack> activeTrack; 4223 Vector< sp<EffectChain> > effectChains; 4224 4225 nsecs_t lastWarning = 0; 4226 4227 acquireWakeLock(); 4228 4229 // start recording 4230 while (!exitPending()) { 4231 4232 processConfigEvents(); 4233 4234 { // scope for mLock 4235 Mutex::Autolock _l(mLock); 4236 checkForNewParameters_l(); 4237 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4238 if (!mStandby) { 4239 mInput->stream->common.standby(&mInput->stream->common); 4240 mStandby = true; 4241 } 4242 4243 if (exitPending()) break; 4244 4245 releaseWakeLock_l(); 4246 ALOGV("RecordThread: loop stopping"); 4247 // go to sleep 4248 mWaitWorkCV.wait(mLock); 4249 ALOGV("RecordThread: loop starting"); 4250 acquireWakeLock_l(); 4251 continue; 4252 } 4253 if (mActiveTrack != 0) { 4254 if (mActiveTrack->mState == TrackBase::PAUSING) { 4255 if (!mStandby) { 4256 mInput->stream->common.standby(&mInput->stream->common); 4257 mStandby = true; 4258 } 4259 mActiveTrack.clear(); 4260 mStartStopCond.broadcast(); 4261 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4262 if (mReqChannelCount != mActiveTrack->channelCount()) { 4263 mActiveTrack.clear(); 4264 mStartStopCond.broadcast(); 4265 } else if (mBytesRead != 0) { 4266 // record start succeeds only if first read from audio input 4267 // succeeds 4268 if (mBytesRead > 0) { 4269 mActiveTrack->mState = TrackBase::ACTIVE; 4270 } else { 4271 mActiveTrack.clear(); 4272 } 4273 mStartStopCond.broadcast(); 4274 } 4275 mStandby = false; 4276 } 4277 } 4278 lockEffectChains_l(effectChains); 4279 } 4280 4281 if (mActiveTrack != 0) { 4282 if (mActiveTrack->mState != TrackBase::ACTIVE && 4283 mActiveTrack->mState != TrackBase::RESUMING) { 4284 unlockEffectChains(effectChains); 4285 usleep(kRecordThreadSleepUs); 4286 continue; 4287 } 4288 for (size_t i = 0; i < effectChains.size(); i ++) { 4289 effectChains[i]->process_l(); 4290 } 4291 4292 buffer.frameCount = mFrameCount; 4293 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4294 size_t framesOut = buffer.frameCount; 4295 if (mResampler == 0) { 4296 // no resampling 4297 while (framesOut) { 4298 size_t framesIn = mFrameCount - mRsmpInIndex; 4299 if (framesIn) { 4300 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4301 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4302 if (framesIn > framesOut) 4303 framesIn = framesOut; 4304 mRsmpInIndex += framesIn; 4305 framesOut -= framesIn; 4306 if ((int)mChannelCount == mReqChannelCount || 4307 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4308 memcpy(dst, src, framesIn * mFrameSize); 4309 } else { 4310 int16_t *src16 = (int16_t *)src; 4311 int16_t *dst16 = (int16_t *)dst; 4312 if (mChannelCount == 1) { 4313 while (framesIn--) { 4314 *dst16++ = *src16; 4315 *dst16++ = *src16++; 4316 } 4317 } else { 4318 while (framesIn--) { 4319 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4320 src16 += 2; 4321 } 4322 } 4323 } 4324 } 4325 if (framesOut && mFrameCount == mRsmpInIndex) { 4326 if (framesOut == mFrameCount && 4327 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4328 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4329 framesOut = 0; 4330 } else { 4331 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4332 mRsmpInIndex = 0; 4333 } 4334 if (mBytesRead < 0) { 4335 LOGE("Error reading audio input"); 4336 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4337 // Force input into standby so that it tries to 4338 // recover at next read attempt 4339 mInput->stream->common.standby(&mInput->stream->common); 4340 usleep(kRecordThreadSleepUs); 4341 } 4342 mRsmpInIndex = mFrameCount; 4343 framesOut = 0; 4344 buffer.frameCount = 0; 4345 } 4346 } 4347 } 4348 } else { 4349 // resampling 4350 4351 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4352 // alter output frame count as if we were expecting stereo samples 4353 if (mChannelCount == 1 && mReqChannelCount == 1) { 4354 framesOut >>= 1; 4355 } 4356 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4357 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4358 // are 32 bit aligned which should be always true. 4359 if (mChannelCount == 2 && mReqChannelCount == 1) { 4360 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4361 // the resampler always outputs stereo samples: do post stereo to mono conversion 4362 int16_t *src = (int16_t *)mRsmpOutBuffer; 4363 int16_t *dst = buffer.i16; 4364 while (framesOut--) { 4365 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4366 src += 2; 4367 } 4368 } else { 4369 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4370 } 4371 4372 } 4373 mActiveTrack->releaseBuffer(&buffer); 4374 mActiveTrack->overflow(); 4375 } 4376 // client isn't retrieving buffers fast enough 4377 else { 4378 if (!mActiveTrack->setOverflow()) { 4379 nsecs_t now = systemTime(); 4380 if ((now - lastWarning) > kWarningThrottle) { 4381 LOGW("RecordThread: buffer overflow"); 4382 lastWarning = now; 4383 } 4384 } 4385 // Release the processor for a while before asking for a new buffer. 4386 // This will give the application more chance to read from the buffer and 4387 // clear the overflow. 4388 usleep(kRecordThreadSleepUs); 4389 } 4390 } 4391 // enable changes in effect chain 4392 unlockEffectChains(effectChains); 4393 effectChains.clear(); 4394 } 4395 4396 if (!mStandby) { 4397 mInput->stream->common.standby(&mInput->stream->common); 4398 } 4399 mActiveTrack.clear(); 4400 4401 mStartStopCond.broadcast(); 4402 4403 releaseWakeLock(); 4404 4405 ALOGV("RecordThread %p exiting", this); 4406 return false; 4407} 4408 4409 4410sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4411 const sp<AudioFlinger::Client>& client, 4412 uint32_t sampleRate, 4413 int format, 4414 int channelMask, 4415 int frameCount, 4416 uint32_t flags, 4417 int sessionId, 4418 status_t *status) 4419{ 4420 sp<RecordTrack> track; 4421 status_t lStatus; 4422 4423 lStatus = initCheck(); 4424 if (lStatus != NO_ERROR) { 4425 LOGE("Audio driver not initialized."); 4426 goto Exit; 4427 } 4428 4429 { // scope for mLock 4430 Mutex::Autolock _l(mLock); 4431 4432 track = new RecordTrack(this, client, sampleRate, 4433 format, channelMask, frameCount, flags, sessionId); 4434 4435 if (track->getCblk() == NULL) { 4436 lStatus = NO_MEMORY; 4437 goto Exit; 4438 } 4439 4440 mTrack = track.get(); 4441 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4442 bool suspend = audio_is_bluetooth_sco_device( 4443 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4444 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4445 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4446 } 4447 lStatus = NO_ERROR; 4448 4449Exit: 4450 if (status) { 4451 *status = lStatus; 4452 } 4453 return track; 4454} 4455 4456status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4457{ 4458 ALOGV("RecordThread::start"); 4459 sp <ThreadBase> strongMe = this; 4460 status_t status = NO_ERROR; 4461 { 4462 AutoMutex lock(&mLock); 4463 if (mActiveTrack != 0) { 4464 if (recordTrack != mActiveTrack.get()) { 4465 status = -EBUSY; 4466 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4467 mActiveTrack->mState = TrackBase::ACTIVE; 4468 } 4469 return status; 4470 } 4471 4472 recordTrack->mState = TrackBase::IDLE; 4473 mActiveTrack = recordTrack; 4474 mLock.unlock(); 4475 status_t status = AudioSystem::startInput(mId); 4476 mLock.lock(); 4477 if (status != NO_ERROR) { 4478 mActiveTrack.clear(); 4479 return status; 4480 } 4481 mRsmpInIndex = mFrameCount; 4482 mBytesRead = 0; 4483 if (mResampler != NULL) { 4484 mResampler->reset(); 4485 } 4486 mActiveTrack->mState = TrackBase::RESUMING; 4487 // signal thread to start 4488 ALOGV("Signal record thread"); 4489 mWaitWorkCV.signal(); 4490 // do not wait for mStartStopCond if exiting 4491 if (mExiting) { 4492 mActiveTrack.clear(); 4493 status = INVALID_OPERATION; 4494 goto startError; 4495 } 4496 mStartStopCond.wait(mLock); 4497 if (mActiveTrack == 0) { 4498 ALOGV("Record failed to start"); 4499 status = BAD_VALUE; 4500 goto startError; 4501 } 4502 ALOGV("Record started OK"); 4503 return status; 4504 } 4505startError: 4506 AudioSystem::stopInput(mId); 4507 return status; 4508} 4509 4510void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4511 ALOGV("RecordThread::stop"); 4512 sp <ThreadBase> strongMe = this; 4513 { 4514 AutoMutex lock(&mLock); 4515 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4516 mActiveTrack->mState = TrackBase::PAUSING; 4517 // do not wait for mStartStopCond if exiting 4518 if (mExiting) { 4519 return; 4520 } 4521 mStartStopCond.wait(mLock); 4522 // if we have been restarted, recordTrack == mActiveTrack.get() here 4523 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4524 mLock.unlock(); 4525 AudioSystem::stopInput(mId); 4526 mLock.lock(); 4527 ALOGV("Record stopped OK"); 4528 } 4529 } 4530 } 4531} 4532 4533status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4534{ 4535 const size_t SIZE = 256; 4536 char buffer[SIZE]; 4537 String8 result; 4538 pid_t pid = 0; 4539 4540 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4541 result.append(buffer); 4542 4543 if (mActiveTrack != 0) { 4544 result.append("Active Track:\n"); 4545 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4546 mActiveTrack->dump(buffer, SIZE); 4547 result.append(buffer); 4548 4549 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4550 result.append(buffer); 4551 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4552 result.append(buffer); 4553 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4554 result.append(buffer); 4555 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4556 result.append(buffer); 4557 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4558 result.append(buffer); 4559 4560 4561 } else { 4562 result.append("No record client\n"); 4563 } 4564 write(fd, result.string(), result.size()); 4565 4566 dumpBase(fd, args); 4567 dumpEffectChains(fd, args); 4568 4569 return NO_ERROR; 4570} 4571 4572status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4573{ 4574 size_t framesReq = buffer->frameCount; 4575 size_t framesReady = mFrameCount - mRsmpInIndex; 4576 int channelCount; 4577 4578 if (framesReady == 0) { 4579 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4580 if (mBytesRead < 0) { 4581 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4582 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4583 // Force input into standby so that it tries to 4584 // recover at next read attempt 4585 mInput->stream->common.standby(&mInput->stream->common); 4586 usleep(kRecordThreadSleepUs); 4587 } 4588 buffer->raw = 0; 4589 buffer->frameCount = 0; 4590 return NOT_ENOUGH_DATA; 4591 } 4592 mRsmpInIndex = 0; 4593 framesReady = mFrameCount; 4594 } 4595 4596 if (framesReq > framesReady) { 4597 framesReq = framesReady; 4598 } 4599 4600 if (mChannelCount == 1 && mReqChannelCount == 2) { 4601 channelCount = 1; 4602 } else { 4603 channelCount = 2; 4604 } 4605 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4606 buffer->frameCount = framesReq; 4607 return NO_ERROR; 4608} 4609 4610void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4611{ 4612 mRsmpInIndex += buffer->frameCount; 4613 buffer->frameCount = 0; 4614} 4615 4616bool AudioFlinger::RecordThread::checkForNewParameters_l() 4617{ 4618 bool reconfig = false; 4619 4620 while (!mNewParameters.isEmpty()) { 4621 status_t status = NO_ERROR; 4622 String8 keyValuePair = mNewParameters[0]; 4623 AudioParameter param = AudioParameter(keyValuePair); 4624 int value; 4625 int reqFormat = mFormat; 4626 int reqSamplingRate = mReqSampleRate; 4627 int reqChannelCount = mReqChannelCount; 4628 4629 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4630 reqSamplingRate = value; 4631 reconfig = true; 4632 } 4633 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4634 reqFormat = value; 4635 reconfig = true; 4636 } 4637 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4638 reqChannelCount = popcount(value); 4639 reconfig = true; 4640 } 4641 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4642 // do not accept frame count changes if tracks are open as the track buffer 4643 // size depends on frame count and correct behavior would not be garantied 4644 // if frame count is changed after track creation 4645 if (mActiveTrack != 0) { 4646 status = INVALID_OPERATION; 4647 } else { 4648 reconfig = true; 4649 } 4650 } 4651 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4652 // forward device change to effects that have requested to be 4653 // aware of attached audio device. 4654 for (size_t i = 0; i < mEffectChains.size(); i++) { 4655 mEffectChains[i]->setDevice_l(value); 4656 } 4657 // store input device and output device but do not forward output device to audio HAL. 4658 // Note that status is ignored by the caller for output device 4659 // (see AudioFlinger::setParameters() 4660 if (value & AUDIO_DEVICE_OUT_ALL) { 4661 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4662 status = BAD_VALUE; 4663 } else { 4664 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4665 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4666 if (mTrack != NULL) { 4667 bool suspend = audio_is_bluetooth_sco_device( 4668 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4669 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4670 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4671 } 4672 } 4673 mDevice |= (uint32_t)value; 4674 } 4675 if (status == NO_ERROR) { 4676 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4677 if (status == INVALID_OPERATION) { 4678 mInput->stream->common.standby(&mInput->stream->common); 4679 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4680 } 4681 if (reconfig) { 4682 if (status == BAD_VALUE && 4683 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4684 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4685 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4686 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4687 (reqChannelCount < 3)) { 4688 status = NO_ERROR; 4689 } 4690 if (status == NO_ERROR) { 4691 readInputParameters(); 4692 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4693 } 4694 } 4695 } 4696 4697 mNewParameters.removeAt(0); 4698 4699 mParamStatus = status; 4700 mParamCond.signal(); 4701 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4702 // already timed out waiting for the status and will never signal the condition. 4703 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout); 4704 } 4705 return reconfig; 4706} 4707 4708String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4709{ 4710 char *s; 4711 String8 out_s8 = String8(); 4712 4713 Mutex::Autolock _l(mLock); 4714 if (initCheck() != NO_ERROR) { 4715 return out_s8; 4716 } 4717 4718 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4719 out_s8 = String8(s); 4720 free(s); 4721 return out_s8; 4722} 4723 4724void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4725 AudioSystem::OutputDescriptor desc; 4726 void *param2 = 0; 4727 4728 switch (event) { 4729 case AudioSystem::INPUT_OPENED: 4730 case AudioSystem::INPUT_CONFIG_CHANGED: 4731 desc.channels = mChannelMask; 4732 desc.samplingRate = mSampleRate; 4733 desc.format = mFormat; 4734 desc.frameCount = mFrameCount; 4735 desc.latency = 0; 4736 param2 = &desc; 4737 break; 4738 4739 case AudioSystem::INPUT_CLOSED: 4740 default: 4741 break; 4742 } 4743 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4744} 4745 4746void AudioFlinger::RecordThread::readInputParameters() 4747{ 4748 if (mRsmpInBuffer) delete mRsmpInBuffer; 4749 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4750 if (mResampler) delete mResampler; 4751 mResampler = 0; 4752 4753 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4754 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4755 mChannelCount = (uint16_t)popcount(mChannelMask); 4756 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4757 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4758 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4759 mFrameCount = mInputBytes / mFrameSize; 4760 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4761 4762 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4763 { 4764 int channelCount; 4765 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4766 // stereo to mono post process as the resampler always outputs stereo. 4767 if (mChannelCount == 1 && mReqChannelCount == 2) { 4768 channelCount = 1; 4769 } else { 4770 channelCount = 2; 4771 } 4772 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4773 mResampler->setSampleRate(mSampleRate); 4774 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4775 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4776 4777 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4778 if (mChannelCount == 1 && mReqChannelCount == 1) { 4779 mFrameCount >>= 1; 4780 } 4781 4782 } 4783 mRsmpInIndex = mFrameCount; 4784} 4785 4786unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4787{ 4788 Mutex::Autolock _l(mLock); 4789 if (initCheck() != NO_ERROR) { 4790 return 0; 4791 } 4792 4793 return mInput->stream->get_input_frames_lost(mInput->stream); 4794} 4795 4796uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4797{ 4798 Mutex::Autolock _l(mLock); 4799 uint32_t result = 0; 4800 if (getEffectChain_l(sessionId) != 0) { 4801 result = EFFECT_SESSION; 4802 } 4803 4804 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4805 result |= TRACK_SESSION; 4806 } 4807 4808 return result; 4809} 4810 4811AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4812{ 4813 Mutex::Autolock _l(mLock); 4814 return mTrack; 4815} 4816 4817AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() 4818{ 4819 Mutex::Autolock _l(mLock); 4820 return mInput; 4821} 4822 4823AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4824{ 4825 Mutex::Autolock _l(mLock); 4826 AudioStreamIn *input = mInput; 4827 mInput = NULL; 4828 return input; 4829} 4830 4831// this method must always be called either with ThreadBase mLock held or inside the thread loop 4832audio_stream_t* AudioFlinger::RecordThread::stream() 4833{ 4834 if (mInput == NULL) { 4835 return NULL; 4836 } 4837 return &mInput->stream->common; 4838} 4839 4840 4841// ---------------------------------------------------------------------------- 4842 4843int AudioFlinger::openOutput(uint32_t *pDevices, 4844 uint32_t *pSamplingRate, 4845 uint32_t *pFormat, 4846 uint32_t *pChannels, 4847 uint32_t *pLatencyMs, 4848 uint32_t flags) 4849{ 4850 status_t status; 4851 PlaybackThread *thread = NULL; 4852 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4853 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4854 uint32_t format = pFormat ? *pFormat : 0; 4855 uint32_t channels = pChannels ? *pChannels : 0; 4856 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4857 audio_stream_out_t *outStream; 4858 audio_hw_device_t *outHwDev; 4859 4860 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4861 pDevices ? *pDevices : 0, 4862 samplingRate, 4863 format, 4864 channels, 4865 flags); 4866 4867 if (pDevices == NULL || *pDevices == 0) { 4868 return 0; 4869 } 4870 4871 Mutex::Autolock _l(mLock); 4872 4873 outHwDev = findSuitableHwDev_l(*pDevices); 4874 if (outHwDev == NULL) 4875 return 0; 4876 4877 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4878 &channels, &samplingRate, &outStream); 4879 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4880 outStream, 4881 samplingRate, 4882 format, 4883 channels, 4884 status); 4885 4886 mHardwareStatus = AUDIO_HW_IDLE; 4887 if (outStream != NULL) { 4888 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4889 int id = nextUniqueId(); 4890 4891 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4892 (format != AUDIO_FORMAT_PCM_16_BIT) || 4893 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4894 thread = new DirectOutputThread(this, output, id, *pDevices); 4895 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4896 } else { 4897 thread = new MixerThread(this, output, id, *pDevices); 4898 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4899 } 4900 mPlaybackThreads.add(id, thread); 4901 4902 if (pSamplingRate) *pSamplingRate = samplingRate; 4903 if (pFormat) *pFormat = format; 4904 if (pChannels) *pChannels = channels; 4905 if (pLatencyMs) *pLatencyMs = thread->latency(); 4906 4907 // notify client processes of the new output creation 4908 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4909 return id; 4910 } 4911 4912 return 0; 4913} 4914 4915int AudioFlinger::openDuplicateOutput(int output1, int output2) 4916{ 4917 Mutex::Autolock _l(mLock); 4918 MixerThread *thread1 = checkMixerThread_l(output1); 4919 MixerThread *thread2 = checkMixerThread_l(output2); 4920 4921 if (thread1 == NULL || thread2 == NULL) { 4922 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4923 return 0; 4924 } 4925 4926 int id = nextUniqueId(); 4927 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4928 thread->addOutputTrack(thread2); 4929 mPlaybackThreads.add(id, thread); 4930 // notify client processes of the new output creation 4931 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4932 return id; 4933} 4934 4935status_t AudioFlinger::closeOutput(int output) 4936{ 4937 // keep strong reference on the playback thread so that 4938 // it is not destroyed while exit() is executed 4939 sp <PlaybackThread> thread; 4940 { 4941 Mutex::Autolock _l(mLock); 4942 thread = checkPlaybackThread_l(output); 4943 if (thread == NULL) { 4944 return BAD_VALUE; 4945 } 4946 4947 ALOGV("closeOutput() %d", output); 4948 4949 if (thread->type() == ThreadBase::MIXER) { 4950 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4951 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4952 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4953 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4954 } 4955 } 4956 } 4957 void *param2 = 0; 4958 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4959 mPlaybackThreads.removeItem(output); 4960 } 4961 thread->exit(); 4962 4963 if (thread->type() != ThreadBase::DUPLICATING) { 4964 AudioStreamOut *out = thread->clearOutput(); 4965 // from now on thread->mOutput is NULL 4966 out->hwDev->close_output_stream(out->hwDev, out->stream); 4967 delete out; 4968 } 4969 return NO_ERROR; 4970} 4971 4972status_t AudioFlinger::suspendOutput(int output) 4973{ 4974 Mutex::Autolock _l(mLock); 4975 PlaybackThread *thread = checkPlaybackThread_l(output); 4976 4977 if (thread == NULL) { 4978 return BAD_VALUE; 4979 } 4980 4981 ALOGV("suspendOutput() %d", output); 4982 thread->suspend(); 4983 4984 return NO_ERROR; 4985} 4986 4987status_t AudioFlinger::restoreOutput(int output) 4988{ 4989 Mutex::Autolock _l(mLock); 4990 PlaybackThread *thread = checkPlaybackThread_l(output); 4991 4992 if (thread == NULL) { 4993 return BAD_VALUE; 4994 } 4995 4996 ALOGV("restoreOutput() %d", output); 4997 4998 thread->restore(); 4999 5000 return NO_ERROR; 5001} 5002 5003int AudioFlinger::openInput(uint32_t *pDevices, 5004 uint32_t *pSamplingRate, 5005 uint32_t *pFormat, 5006 uint32_t *pChannels, 5007 uint32_t acoustics) 5008{ 5009 status_t status; 5010 RecordThread *thread = NULL; 5011 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5012 uint32_t format = pFormat ? *pFormat : 0; 5013 uint32_t channels = pChannels ? *pChannels : 0; 5014 uint32_t reqSamplingRate = samplingRate; 5015 uint32_t reqFormat = format; 5016 uint32_t reqChannels = channels; 5017 audio_stream_in_t *inStream; 5018 audio_hw_device_t *inHwDev; 5019 5020 if (pDevices == NULL || *pDevices == 0) { 5021 return 0; 5022 } 5023 5024 Mutex::Autolock _l(mLock); 5025 5026 inHwDev = findSuitableHwDev_l(*pDevices); 5027 if (inHwDev == NULL) 5028 return 0; 5029 5030 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5031 &channels, &samplingRate, 5032 (audio_in_acoustics_t)acoustics, 5033 &inStream); 5034 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5035 inStream, 5036 samplingRate, 5037 format, 5038 channels, 5039 acoustics, 5040 status); 5041 5042 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5043 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5044 // or stereo to mono conversions on 16 bit PCM inputs. 5045 if (inStream == NULL && status == BAD_VALUE && 5046 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5047 (samplingRate <= 2 * reqSamplingRate) && 5048 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5049 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5050 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 5051 &channels, &samplingRate, 5052 (audio_in_acoustics_t)acoustics, 5053 &inStream); 5054 } 5055 5056 if (inStream != NULL) { 5057 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5058 5059 int id = nextUniqueId(); 5060 // Start record thread 5061 // RecorThread require both input and output device indication to forward to audio 5062 // pre processing modules 5063 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5064 thread = new RecordThread(this, 5065 input, 5066 reqSamplingRate, 5067 reqChannels, 5068 id, 5069 device); 5070 mRecordThreads.add(id, thread); 5071 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5072 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5073 if (pFormat) *pFormat = format; 5074 if (pChannels) *pChannels = reqChannels; 5075 5076 input->stream->common.standby(&input->stream->common); 5077 5078 // notify client processes of the new input creation 5079 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5080 return id; 5081 } 5082 5083 return 0; 5084} 5085 5086status_t AudioFlinger::closeInput(int input) 5087{ 5088 // keep strong reference on the record thread so that 5089 // it is not destroyed while exit() is executed 5090 sp <RecordThread> thread; 5091 { 5092 Mutex::Autolock _l(mLock); 5093 thread = checkRecordThread_l(input); 5094 if (thread == NULL) { 5095 return BAD_VALUE; 5096 } 5097 5098 ALOGV("closeInput() %d", input); 5099 void *param2 = 0; 5100 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5101 mRecordThreads.removeItem(input); 5102 } 5103 thread->exit(); 5104 5105 AudioStreamIn *in = thread->clearInput(); 5106 // from now on thread->mInput is NULL 5107 in->hwDev->close_input_stream(in->hwDev, in->stream); 5108 delete in; 5109 5110 return NO_ERROR; 5111} 5112 5113status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 5114{ 5115 Mutex::Autolock _l(mLock); 5116 MixerThread *dstThread = checkMixerThread_l(output); 5117 if (dstThread == NULL) { 5118 LOGW("setStreamOutput() bad output id %d", output); 5119 return BAD_VALUE; 5120 } 5121 5122 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5123 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5124 5125 dstThread->setStreamValid(stream, true); 5126 5127 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5128 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5129 if (thread != dstThread && 5130 thread->type() != ThreadBase::DIRECT) { 5131 MixerThread *srcThread = (MixerThread *)thread; 5132 srcThread->setStreamValid(stream, false); 5133 srcThread->invalidateTracks(stream); 5134 } 5135 } 5136 5137 return NO_ERROR; 5138} 5139 5140 5141int AudioFlinger::newAudioSessionId() 5142{ 5143 return nextUniqueId(); 5144} 5145 5146void AudioFlinger::acquireAudioSessionId(int audioSession) 5147{ 5148 Mutex::Autolock _l(mLock); 5149 int caller = IPCThreadState::self()->getCallingPid(); 5150 ALOGV("acquiring %d from %d", audioSession, caller); 5151 int num = mAudioSessionRefs.size(); 5152 for (int i = 0; i< num; i++) { 5153 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5154 if (ref->sessionid == audioSession && ref->pid == caller) { 5155 ref->cnt++; 5156 ALOGV(" incremented refcount to %d", ref->cnt); 5157 return; 5158 } 5159 } 5160 AudioSessionRef *ref = new AudioSessionRef(); 5161 ref->sessionid = audioSession; 5162 ref->pid = caller; 5163 ref->cnt = 1; 5164 mAudioSessionRefs.push(ref); 5165 ALOGV(" added new entry for %d", ref->sessionid); 5166} 5167 5168void AudioFlinger::releaseAudioSessionId(int audioSession) 5169{ 5170 Mutex::Autolock _l(mLock); 5171 int caller = IPCThreadState::self()->getCallingPid(); 5172 ALOGV("releasing %d from %d", audioSession, caller); 5173 int num = mAudioSessionRefs.size(); 5174 for (int i = 0; i< num; i++) { 5175 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5176 if (ref->sessionid == audioSession && ref->pid == caller) { 5177 ref->cnt--; 5178 ALOGV(" decremented refcount to %d", ref->cnt); 5179 if (ref->cnt == 0) { 5180 mAudioSessionRefs.removeAt(i); 5181 delete ref; 5182 purgeStaleEffects_l(); 5183 } 5184 return; 5185 } 5186 } 5187 LOGW("session id %d not found for pid %d", audioSession, caller); 5188} 5189 5190void AudioFlinger::purgeStaleEffects_l() { 5191 5192 ALOGV("purging stale effects"); 5193 5194 Vector< sp<EffectChain> > chains; 5195 5196 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5197 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5198 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5199 sp<EffectChain> ec = t->mEffectChains[j]; 5200 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5201 chains.push(ec); 5202 } 5203 } 5204 } 5205 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5206 sp<RecordThread> t = mRecordThreads.valueAt(i); 5207 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5208 sp<EffectChain> ec = t->mEffectChains[j]; 5209 chains.push(ec); 5210 } 5211 } 5212 5213 for (size_t i = 0; i < chains.size(); i++) { 5214 sp<EffectChain> ec = chains[i]; 5215 int sessionid = ec->sessionId(); 5216 sp<ThreadBase> t = ec->mThread.promote(); 5217 if (t == 0) { 5218 continue; 5219 } 5220 size_t numsessionrefs = mAudioSessionRefs.size(); 5221 bool found = false; 5222 for (size_t k = 0; k < numsessionrefs; k++) { 5223 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5224 if (ref->sessionid == sessionid) { 5225 ALOGV(" session %d still exists for %d with %d refs", 5226 sessionid, ref->pid, ref->cnt); 5227 found = true; 5228 break; 5229 } 5230 } 5231 if (!found) { 5232 // remove all effects from the chain 5233 while (ec->mEffects.size()) { 5234 sp<EffectModule> effect = ec->mEffects[0]; 5235 effect->unPin(); 5236 Mutex::Autolock _l (t->mLock); 5237 t->removeEffect_l(effect); 5238 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5239 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5240 if (handle != 0) { 5241 handle->mEffect.clear(); 5242 if (handle->mHasControl && handle->mEnabled) { 5243 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5244 } 5245 } 5246 } 5247 AudioSystem::unregisterEffect(effect->id()); 5248 } 5249 } 5250 } 5251 return; 5252} 5253 5254// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5255AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5256{ 5257 PlaybackThread *thread = NULL; 5258 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5259 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5260 } 5261 return thread; 5262} 5263 5264// checkMixerThread_l() must be called with AudioFlinger::mLock held 5265AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5266{ 5267 PlaybackThread *thread = checkPlaybackThread_l(output); 5268 if (thread != NULL) { 5269 if (thread->type() == ThreadBase::DIRECT) { 5270 thread = NULL; 5271 } 5272 } 5273 return (MixerThread *)thread; 5274} 5275 5276// checkRecordThread_l() must be called with AudioFlinger::mLock held 5277AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5278{ 5279 RecordThread *thread = NULL; 5280 if (mRecordThreads.indexOfKey(input) >= 0) { 5281 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5282 } 5283 return thread; 5284} 5285 5286uint32_t AudioFlinger::nextUniqueId() 5287{ 5288 return android_atomic_inc(&mNextUniqueId); 5289} 5290 5291AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5292{ 5293 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5294 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5295 AudioStreamOut *output = thread->getOutput(); 5296 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5297 return thread; 5298 } 5299 } 5300 return NULL; 5301} 5302 5303uint32_t AudioFlinger::primaryOutputDevice_l() 5304{ 5305 PlaybackThread *thread = primaryPlaybackThread_l(); 5306 5307 if (thread == NULL) { 5308 return 0; 5309 } 5310 5311 return thread->device(); 5312} 5313 5314 5315// ---------------------------------------------------------------------------- 5316// Effect management 5317// ---------------------------------------------------------------------------- 5318 5319 5320status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5321{ 5322 Mutex::Autolock _l(mLock); 5323 return EffectQueryNumberEffects(numEffects); 5324} 5325 5326status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5327{ 5328 Mutex::Autolock _l(mLock); 5329 return EffectQueryEffect(index, descriptor); 5330} 5331 5332status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5333{ 5334 Mutex::Autolock _l(mLock); 5335 return EffectGetDescriptor(pUuid, descriptor); 5336} 5337 5338 5339sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5340 effect_descriptor_t *pDesc, 5341 const sp<IEffectClient>& effectClient, 5342 int32_t priority, 5343 int io, 5344 int sessionId, 5345 status_t *status, 5346 int *id, 5347 int *enabled) 5348{ 5349 status_t lStatus = NO_ERROR; 5350 sp<EffectHandle> handle; 5351 effect_descriptor_t desc; 5352 sp<Client> client; 5353 wp<Client> wclient; 5354 5355 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5356 pid, effectClient.get(), priority, sessionId, io); 5357 5358 if (pDesc == NULL) { 5359 lStatus = BAD_VALUE; 5360 goto Exit; 5361 } 5362 5363 // check audio settings permission for global effects 5364 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5365 lStatus = PERMISSION_DENIED; 5366 goto Exit; 5367 } 5368 5369 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5370 // that can only be created by audio policy manager (running in same process) 5371 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5372 lStatus = PERMISSION_DENIED; 5373 goto Exit; 5374 } 5375 5376 if (io == 0) { 5377 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5378 // output must be specified by AudioPolicyManager when using session 5379 // AUDIO_SESSION_OUTPUT_STAGE 5380 lStatus = BAD_VALUE; 5381 goto Exit; 5382 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5383 // if the output returned by getOutputForEffect() is removed before we lock the 5384 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5385 // and we will exit safely 5386 io = AudioSystem::getOutputForEffect(&desc); 5387 } 5388 } 5389 5390 { 5391 Mutex::Autolock _l(mLock); 5392 5393 5394 if (!EffectIsNullUuid(&pDesc->uuid)) { 5395 // if uuid is specified, request effect descriptor 5396 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5397 if (lStatus < 0) { 5398 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5399 goto Exit; 5400 } 5401 } else { 5402 // if uuid is not specified, look for an available implementation 5403 // of the required type in effect factory 5404 if (EffectIsNullUuid(&pDesc->type)) { 5405 LOGW("createEffect() no effect type"); 5406 lStatus = BAD_VALUE; 5407 goto Exit; 5408 } 5409 uint32_t numEffects = 0; 5410 effect_descriptor_t d; 5411 d.flags = 0; // prevent compiler warning 5412 bool found = false; 5413 5414 lStatus = EffectQueryNumberEffects(&numEffects); 5415 if (lStatus < 0) { 5416 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5417 goto Exit; 5418 } 5419 for (uint32_t i = 0; i < numEffects; i++) { 5420 lStatus = EffectQueryEffect(i, &desc); 5421 if (lStatus < 0) { 5422 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5423 continue; 5424 } 5425 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5426 // If matching type found save effect descriptor. If the session is 5427 // 0 and the effect is not auxiliary, continue enumeration in case 5428 // an auxiliary version of this effect type is available 5429 found = true; 5430 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5431 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5432 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5433 break; 5434 } 5435 } 5436 } 5437 if (!found) { 5438 lStatus = BAD_VALUE; 5439 LOGW("createEffect() effect not found"); 5440 goto Exit; 5441 } 5442 // For same effect type, chose auxiliary version over insert version if 5443 // connect to output mix (Compliance to OpenSL ES) 5444 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5445 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5446 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5447 } 5448 } 5449 5450 // Do not allow auxiliary effects on a session different from 0 (output mix) 5451 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5452 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5453 lStatus = INVALID_OPERATION; 5454 goto Exit; 5455 } 5456 5457 // check recording permission for visualizer 5458 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5459 !recordingAllowed()) { 5460 lStatus = PERMISSION_DENIED; 5461 goto Exit; 5462 } 5463 5464 // return effect descriptor 5465 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5466 5467 // If output is not specified try to find a matching audio session ID in one of the 5468 // output threads. 5469 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5470 // because of code checking output when entering the function. 5471 // Note: io is never 0 when creating an effect on an input 5472 if (io == 0) { 5473 // look for the thread where the specified audio session is present 5474 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5475 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5476 io = mPlaybackThreads.keyAt(i); 5477 break; 5478 } 5479 } 5480 if (io == 0) { 5481 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5482 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5483 io = mRecordThreads.keyAt(i); 5484 break; 5485 } 5486 } 5487 } 5488 // If no output thread contains the requested session ID, default to 5489 // first output. The effect chain will be moved to the correct output 5490 // thread when a track with the same session ID is created 5491 if (io == 0 && mPlaybackThreads.size()) { 5492 io = mPlaybackThreads.keyAt(0); 5493 } 5494 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5495 } 5496 ThreadBase *thread = checkRecordThread_l(io); 5497 if (thread == NULL) { 5498 thread = checkPlaybackThread_l(io); 5499 if (thread == NULL) { 5500 LOGE("createEffect() unknown output thread"); 5501 lStatus = BAD_VALUE; 5502 goto Exit; 5503 } 5504 } 5505 5506 wclient = mClients.valueFor(pid); 5507 5508 if (wclient != NULL) { 5509 client = wclient.promote(); 5510 } else { 5511 client = new Client(this, pid); 5512 mClients.add(pid, client); 5513 } 5514 5515 // create effect on selected output thread 5516 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5517 &desc, enabled, &lStatus); 5518 if (handle != 0 && id != NULL) { 5519 *id = handle->id(); 5520 } 5521 } 5522 5523Exit: 5524 if(status) { 5525 *status = lStatus; 5526 } 5527 return handle; 5528} 5529 5530status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5531{ 5532 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5533 sessionId, srcOutput, dstOutput); 5534 Mutex::Autolock _l(mLock); 5535 if (srcOutput == dstOutput) { 5536 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 5537 return NO_ERROR; 5538 } 5539 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5540 if (srcThread == NULL) { 5541 LOGW("moveEffects() bad srcOutput %d", srcOutput); 5542 return BAD_VALUE; 5543 } 5544 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5545 if (dstThread == NULL) { 5546 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5547 return BAD_VALUE; 5548 } 5549 5550 Mutex::Autolock _dl(dstThread->mLock); 5551 Mutex::Autolock _sl(srcThread->mLock); 5552 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5553 5554 return NO_ERROR; 5555} 5556 5557// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5558status_t AudioFlinger::moveEffectChain_l(int sessionId, 5559 AudioFlinger::PlaybackThread *srcThread, 5560 AudioFlinger::PlaybackThread *dstThread, 5561 bool reRegister) 5562{ 5563 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5564 sessionId, srcThread, dstThread); 5565 5566 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5567 if (chain == 0) { 5568 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5569 sessionId, srcThread); 5570 return INVALID_OPERATION; 5571 } 5572 5573 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5574 // so that a new chain is created with correct parameters when first effect is added. This is 5575 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5576 // removed. 5577 srcThread->removeEffectChain_l(chain); 5578 5579 // transfer all effects one by one so that new effect chain is created on new thread with 5580 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5581 int dstOutput = dstThread->id(); 5582 sp<EffectChain> dstChain; 5583 uint32_t strategy = 0; // prevent compiler warning 5584 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5585 while (effect != 0) { 5586 srcThread->removeEffect_l(effect); 5587 dstThread->addEffect_l(effect); 5588 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5589 if (effect->state() == EffectModule::ACTIVE || 5590 effect->state() == EffectModule::STOPPING) { 5591 effect->start(); 5592 } 5593 // if the move request is not received from audio policy manager, the effect must be 5594 // re-registered with the new strategy and output 5595 if (dstChain == 0) { 5596 dstChain = effect->chain().promote(); 5597 if (dstChain == 0) { 5598 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5599 srcThread->addEffect_l(effect); 5600 return NO_INIT; 5601 } 5602 strategy = dstChain->strategy(); 5603 } 5604 if (reRegister) { 5605 AudioSystem::unregisterEffect(effect->id()); 5606 AudioSystem::registerEffect(&effect->desc(), 5607 dstOutput, 5608 strategy, 5609 sessionId, 5610 effect->id()); 5611 } 5612 effect = chain->getEffectFromId_l(0); 5613 } 5614 5615 return NO_ERROR; 5616} 5617 5618 5619// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5620sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5621 const sp<AudioFlinger::Client>& client, 5622 const sp<IEffectClient>& effectClient, 5623 int32_t priority, 5624 int sessionId, 5625 effect_descriptor_t *desc, 5626 int *enabled, 5627 status_t *status 5628 ) 5629{ 5630 sp<EffectModule> effect; 5631 sp<EffectHandle> handle; 5632 status_t lStatus; 5633 sp<EffectChain> chain; 5634 bool chainCreated = false; 5635 bool effectCreated = false; 5636 bool effectRegistered = false; 5637 5638 lStatus = initCheck(); 5639 if (lStatus != NO_ERROR) { 5640 LOGW("createEffect_l() Audio driver not initialized."); 5641 goto Exit; 5642 } 5643 5644 // Do not allow effects with session ID 0 on direct output or duplicating threads 5645 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5646 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5647 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5648 desc->name, sessionId); 5649 lStatus = BAD_VALUE; 5650 goto Exit; 5651 } 5652 // Only Pre processor effects are allowed on input threads and only on input threads 5653 if ((mType == RECORD && 5654 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5655 (mType != RECORD && 5656 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5657 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5658 desc->name, desc->flags, mType); 5659 lStatus = BAD_VALUE; 5660 goto Exit; 5661 } 5662 5663 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5664 5665 { // scope for mLock 5666 Mutex::Autolock _l(mLock); 5667 5668 // check for existing effect chain with the requested audio session 5669 chain = getEffectChain_l(sessionId); 5670 if (chain == 0) { 5671 // create a new chain for this session 5672 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5673 chain = new EffectChain(this, sessionId); 5674 addEffectChain_l(chain); 5675 chain->setStrategy(getStrategyForSession_l(sessionId)); 5676 chainCreated = true; 5677 } else { 5678 effect = chain->getEffectFromDesc_l(desc); 5679 } 5680 5681 ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5682 5683 if (effect == 0) { 5684 int id = mAudioFlinger->nextUniqueId(); 5685 // Check CPU and memory usage 5686 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5687 if (lStatus != NO_ERROR) { 5688 goto Exit; 5689 } 5690 effectRegistered = true; 5691 // create a new effect module if none present in the chain 5692 effect = new EffectModule(this, chain, desc, id, sessionId); 5693 lStatus = effect->status(); 5694 if (lStatus != NO_ERROR) { 5695 goto Exit; 5696 } 5697 lStatus = chain->addEffect_l(effect); 5698 if (lStatus != NO_ERROR) { 5699 goto Exit; 5700 } 5701 effectCreated = true; 5702 5703 effect->setDevice(mDevice); 5704 effect->setMode(mAudioFlinger->getMode()); 5705 } 5706 // create effect handle and connect it to effect module 5707 handle = new EffectHandle(effect, client, effectClient, priority); 5708 lStatus = effect->addHandle(handle); 5709 if (enabled) { 5710 *enabled = (int)effect->isEnabled(); 5711 } 5712 } 5713 5714Exit: 5715 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5716 Mutex::Autolock _l(mLock); 5717 if (effectCreated) { 5718 chain->removeEffect_l(effect); 5719 } 5720 if (effectRegistered) { 5721 AudioSystem::unregisterEffect(effect->id()); 5722 } 5723 if (chainCreated) { 5724 removeEffectChain_l(chain); 5725 } 5726 handle.clear(); 5727 } 5728 5729 if(status) { 5730 *status = lStatus; 5731 } 5732 return handle; 5733} 5734 5735sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5736{ 5737 sp<EffectModule> effect; 5738 5739 sp<EffectChain> chain = getEffectChain_l(sessionId); 5740 if (chain != 0) { 5741 effect = chain->getEffectFromId_l(effectId); 5742 } 5743 return effect; 5744} 5745 5746// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5747// PlaybackThread::mLock held 5748status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5749{ 5750 // check for existing effect chain with the requested audio session 5751 int sessionId = effect->sessionId(); 5752 sp<EffectChain> chain = getEffectChain_l(sessionId); 5753 bool chainCreated = false; 5754 5755 if (chain == 0) { 5756 // create a new chain for this session 5757 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5758 chain = new EffectChain(this, sessionId); 5759 addEffectChain_l(chain); 5760 chain->setStrategy(getStrategyForSession_l(sessionId)); 5761 chainCreated = true; 5762 } 5763 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5764 5765 if (chain->getEffectFromId_l(effect->id()) != 0) { 5766 LOGW("addEffect_l() %p effect %s already present in chain %p", 5767 this, effect->desc().name, chain.get()); 5768 return BAD_VALUE; 5769 } 5770 5771 status_t status = chain->addEffect_l(effect); 5772 if (status != NO_ERROR) { 5773 if (chainCreated) { 5774 removeEffectChain_l(chain); 5775 } 5776 return status; 5777 } 5778 5779 effect->setDevice(mDevice); 5780 effect->setMode(mAudioFlinger->getMode()); 5781 return NO_ERROR; 5782} 5783 5784void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5785 5786 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5787 effect_descriptor_t desc = effect->desc(); 5788 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5789 detachAuxEffect_l(effect->id()); 5790 } 5791 5792 sp<EffectChain> chain = effect->chain().promote(); 5793 if (chain != 0) { 5794 // remove effect chain if removing last effect 5795 if (chain->removeEffect_l(effect) == 0) { 5796 removeEffectChain_l(chain); 5797 } 5798 } else { 5799 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5800 } 5801} 5802 5803void AudioFlinger::ThreadBase::lockEffectChains_l( 5804 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5805{ 5806 effectChains = mEffectChains; 5807 for (size_t i = 0; i < mEffectChains.size(); i++) { 5808 mEffectChains[i]->lock(); 5809 } 5810} 5811 5812void AudioFlinger::ThreadBase::unlockEffectChains( 5813 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5814{ 5815 for (size_t i = 0; i < effectChains.size(); i++) { 5816 effectChains[i]->unlock(); 5817 } 5818} 5819 5820sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5821{ 5822 Mutex::Autolock _l(mLock); 5823 return getEffectChain_l(sessionId); 5824} 5825 5826sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5827{ 5828 sp<EffectChain> chain; 5829 5830 size_t size = mEffectChains.size(); 5831 for (size_t i = 0; i < size; i++) { 5832 if (mEffectChains[i]->sessionId() == sessionId) { 5833 chain = mEffectChains[i]; 5834 break; 5835 } 5836 } 5837 return chain; 5838} 5839 5840void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5841{ 5842 Mutex::Autolock _l(mLock); 5843 size_t size = mEffectChains.size(); 5844 for (size_t i = 0; i < size; i++) { 5845 mEffectChains[i]->setMode_l(mode); 5846 } 5847} 5848 5849void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5850 const wp<EffectHandle>& handle, 5851 bool unpiniflast) { 5852 5853 Mutex::Autolock _l(mLock); 5854 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5855 // delete the effect module if removing last handle on it 5856 if (effect->removeHandle(handle) == 0) { 5857 if (!effect->isPinned() || unpiniflast) { 5858 removeEffect_l(effect); 5859 AudioSystem::unregisterEffect(effect->id()); 5860 } 5861 } 5862} 5863 5864status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5865{ 5866 int session = chain->sessionId(); 5867 int16_t *buffer = mMixBuffer; 5868 bool ownsBuffer = false; 5869 5870 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5871 if (session > 0) { 5872 // Only one effect chain can be present in direct output thread and it uses 5873 // the mix buffer as input 5874 if (mType != DIRECT) { 5875 size_t numSamples = mFrameCount * mChannelCount; 5876 buffer = new int16_t[numSamples]; 5877 memset(buffer, 0, numSamples * sizeof(int16_t)); 5878 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5879 ownsBuffer = true; 5880 } 5881 5882 // Attach all tracks with same session ID to this chain. 5883 for (size_t i = 0; i < mTracks.size(); ++i) { 5884 sp<Track> track = mTracks[i]; 5885 if (session == track->sessionId()) { 5886 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5887 track->setMainBuffer(buffer); 5888 chain->incTrackCnt(); 5889 } 5890 } 5891 5892 // indicate all active tracks in the chain 5893 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5894 sp<Track> track = mActiveTracks[i].promote(); 5895 if (track == 0) continue; 5896 if (session == track->sessionId()) { 5897 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5898 chain->incActiveTrackCnt(); 5899 } 5900 } 5901 } 5902 5903 chain->setInBuffer(buffer, ownsBuffer); 5904 chain->setOutBuffer(mMixBuffer); 5905 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5906 // chains list in order to be processed last as it contains output stage effects 5907 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5908 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5909 // after track specific effects and before output stage 5910 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5911 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5912 // Effect chain for other sessions are inserted at beginning of effect 5913 // chains list to be processed before output mix effects. Relative order between other 5914 // sessions is not important 5915 size_t size = mEffectChains.size(); 5916 size_t i = 0; 5917 for (i = 0; i < size; i++) { 5918 if (mEffectChains[i]->sessionId() < session) break; 5919 } 5920 mEffectChains.insertAt(chain, i); 5921 checkSuspendOnAddEffectChain_l(chain); 5922 5923 return NO_ERROR; 5924} 5925 5926size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5927{ 5928 int session = chain->sessionId(); 5929 5930 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5931 5932 for (size_t i = 0; i < mEffectChains.size(); i++) { 5933 if (chain == mEffectChains[i]) { 5934 mEffectChains.removeAt(i); 5935 // detach all active tracks from the chain 5936 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5937 sp<Track> track = mActiveTracks[i].promote(); 5938 if (track == 0) continue; 5939 if (session == track->sessionId()) { 5940 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5941 chain.get(), session); 5942 chain->decActiveTrackCnt(); 5943 } 5944 } 5945 5946 // detach all tracks with same session ID from this chain 5947 for (size_t i = 0; i < mTracks.size(); ++i) { 5948 sp<Track> track = mTracks[i]; 5949 if (session == track->sessionId()) { 5950 track->setMainBuffer(mMixBuffer); 5951 chain->decTrackCnt(); 5952 } 5953 } 5954 break; 5955 } 5956 } 5957 return mEffectChains.size(); 5958} 5959 5960status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5961 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5962{ 5963 Mutex::Autolock _l(mLock); 5964 return attachAuxEffect_l(track, EffectId); 5965} 5966 5967status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5968 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5969{ 5970 status_t status = NO_ERROR; 5971 5972 if (EffectId == 0) { 5973 track->setAuxBuffer(0, NULL); 5974 } else { 5975 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5976 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 5977 if (effect != 0) { 5978 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5979 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5980 } else { 5981 status = INVALID_OPERATION; 5982 } 5983 } else { 5984 status = BAD_VALUE; 5985 } 5986 } 5987 return status; 5988} 5989 5990void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5991{ 5992 for (size_t i = 0; i < mTracks.size(); ++i) { 5993 sp<Track> track = mTracks[i]; 5994 if (track->auxEffectId() == effectId) { 5995 attachAuxEffect_l(track, 0); 5996 } 5997 } 5998} 5999 6000status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6001{ 6002 // only one chain per input thread 6003 if (mEffectChains.size() != 0) { 6004 return INVALID_OPERATION; 6005 } 6006 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6007 6008 chain->setInBuffer(NULL); 6009 chain->setOutBuffer(NULL); 6010 6011 checkSuspendOnAddEffectChain_l(chain); 6012 6013 mEffectChains.add(chain); 6014 6015 return NO_ERROR; 6016} 6017 6018size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6019{ 6020 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6021 LOGW_IF(mEffectChains.size() != 1, 6022 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6023 chain.get(), mEffectChains.size(), this); 6024 if (mEffectChains.size() == 1) { 6025 mEffectChains.removeAt(0); 6026 } 6027 return 0; 6028} 6029 6030// ---------------------------------------------------------------------------- 6031// EffectModule implementation 6032// ---------------------------------------------------------------------------- 6033 6034#undef LOG_TAG 6035#define LOG_TAG "AudioFlinger::EffectModule" 6036 6037AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6038 const wp<AudioFlinger::EffectChain>& chain, 6039 effect_descriptor_t *desc, 6040 int id, 6041 int sessionId) 6042 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6043 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6044{ 6045 ALOGV("Constructor %p", this); 6046 int lStatus; 6047 sp<ThreadBase> thread = mThread.promote(); 6048 if (thread == 0) { 6049 return; 6050 } 6051 6052 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6053 6054 // create effect engine from effect factory 6055 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6056 6057 if (mStatus != NO_ERROR) { 6058 return; 6059 } 6060 lStatus = init(); 6061 if (lStatus < 0) { 6062 mStatus = lStatus; 6063 goto Error; 6064 } 6065 6066 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6067 mPinned = true; 6068 } 6069 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6070 return; 6071Error: 6072 EffectRelease(mEffectInterface); 6073 mEffectInterface = NULL; 6074 ALOGV("Constructor Error %d", mStatus); 6075} 6076 6077AudioFlinger::EffectModule::~EffectModule() 6078{ 6079 ALOGV("Destructor %p", this); 6080 if (mEffectInterface != NULL) { 6081 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6082 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6083 sp<ThreadBase> thread = mThread.promote(); 6084 if (thread != 0) { 6085 audio_stream_t *stream = thread->stream(); 6086 if (stream != NULL) { 6087 stream->remove_audio_effect(stream, mEffectInterface); 6088 } 6089 } 6090 } 6091 // release effect engine 6092 EffectRelease(mEffectInterface); 6093 } 6094} 6095 6096status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6097{ 6098 status_t status; 6099 6100 Mutex::Autolock _l(mLock); 6101 // First handle in mHandles has highest priority and controls the effect module 6102 int priority = handle->priority(); 6103 size_t size = mHandles.size(); 6104 sp<EffectHandle> h; 6105 size_t i; 6106 for (i = 0; i < size; i++) { 6107 h = mHandles[i].promote(); 6108 if (h == 0) continue; 6109 if (h->priority() <= priority) break; 6110 } 6111 // if inserted in first place, move effect control from previous owner to this handle 6112 if (i == 0) { 6113 bool enabled = false; 6114 if (h != 0) { 6115 enabled = h->enabled(); 6116 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6117 } 6118 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6119 status = NO_ERROR; 6120 } else { 6121 status = ALREADY_EXISTS; 6122 } 6123 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6124 mHandles.insertAt(handle, i); 6125 return status; 6126} 6127 6128size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6129{ 6130 Mutex::Autolock _l(mLock); 6131 size_t size = mHandles.size(); 6132 size_t i; 6133 for (i = 0; i < size; i++) { 6134 if (mHandles[i] == handle) break; 6135 } 6136 if (i == size) { 6137 return size; 6138 } 6139 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6140 6141 bool enabled = false; 6142 EffectHandle *hdl = handle.unsafe_get(); 6143 if (hdl) { 6144 ALOGV("removeHandle() unsafe_get OK"); 6145 enabled = hdl->enabled(); 6146 } 6147 mHandles.removeAt(i); 6148 size = mHandles.size(); 6149 // if removed from first place, move effect control from this handle to next in line 6150 if (i == 0 && size != 0) { 6151 sp<EffectHandle> h = mHandles[0].promote(); 6152 if (h != 0) { 6153 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6154 } 6155 } 6156 6157 // Prevent calls to process() and other functions on effect interface from now on. 6158 // The effect engine will be released by the destructor when the last strong reference on 6159 // this object is released which can happen after next process is called. 6160 if (size == 0 && !mPinned) { 6161 mState = DESTROYED; 6162 } 6163 6164 return size; 6165} 6166 6167sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6168{ 6169 Mutex::Autolock _l(mLock); 6170 sp<EffectHandle> handle; 6171 if (mHandles.size() != 0) { 6172 handle = mHandles[0].promote(); 6173 } 6174 return handle; 6175} 6176 6177void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6178{ 6179 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6180 // keep a strong reference on this EffectModule to avoid calling the 6181 // destructor before we exit 6182 sp<EffectModule> keep(this); 6183 { 6184 sp<ThreadBase> thread = mThread.promote(); 6185 if (thread != 0) { 6186 thread->disconnectEffect(keep, handle, unpiniflast); 6187 } 6188 } 6189} 6190 6191void AudioFlinger::EffectModule::updateState() { 6192 Mutex::Autolock _l(mLock); 6193 6194 switch (mState) { 6195 case RESTART: 6196 reset_l(); 6197 // FALL THROUGH 6198 6199 case STARTING: 6200 // clear auxiliary effect input buffer for next accumulation 6201 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6202 memset(mConfig.inputCfg.buffer.raw, 6203 0, 6204 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6205 } 6206 start_l(); 6207 mState = ACTIVE; 6208 break; 6209 case STOPPING: 6210 stop_l(); 6211 mDisableWaitCnt = mMaxDisableWaitCnt; 6212 mState = STOPPED; 6213 break; 6214 case STOPPED: 6215 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6216 // turn off sequence. 6217 if (--mDisableWaitCnt == 0) { 6218 reset_l(); 6219 mState = IDLE; 6220 } 6221 break; 6222 default: //IDLE , ACTIVE, DESTROYED 6223 break; 6224 } 6225} 6226 6227void AudioFlinger::EffectModule::process() 6228{ 6229 Mutex::Autolock _l(mLock); 6230 6231 if (mState == DESTROYED || mEffectInterface == NULL || 6232 mConfig.inputCfg.buffer.raw == NULL || 6233 mConfig.outputCfg.buffer.raw == NULL) { 6234 return; 6235 } 6236 6237 if (isProcessEnabled()) { 6238 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6239 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6240 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 6241 mConfig.inputCfg.buffer.s32, 6242 mConfig.inputCfg.buffer.frameCount/2); 6243 } 6244 6245 // do the actual processing in the effect engine 6246 int ret = (*mEffectInterface)->process(mEffectInterface, 6247 &mConfig.inputCfg.buffer, 6248 &mConfig.outputCfg.buffer); 6249 6250 // force transition to IDLE state when engine is ready 6251 if (mState == STOPPED && ret == -ENODATA) { 6252 mDisableWaitCnt = 1; 6253 } 6254 6255 // clear auxiliary effect input buffer for next accumulation 6256 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6257 memset(mConfig.inputCfg.buffer.raw, 0, 6258 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6259 } 6260 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6261 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6262 // If an insert effect is idle and input buffer is different from output buffer, 6263 // accumulate input onto output 6264 sp<EffectChain> chain = mChain.promote(); 6265 if (chain != 0 && chain->activeTrackCnt() != 0) { 6266 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6267 int16_t *in = mConfig.inputCfg.buffer.s16; 6268 int16_t *out = mConfig.outputCfg.buffer.s16; 6269 for (size_t i = 0; i < frameCnt; i++) { 6270 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6271 } 6272 } 6273 } 6274} 6275 6276void AudioFlinger::EffectModule::reset_l() 6277{ 6278 if (mEffectInterface == NULL) { 6279 return; 6280 } 6281 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6282} 6283 6284status_t AudioFlinger::EffectModule::configure() 6285{ 6286 uint32_t channels; 6287 if (mEffectInterface == NULL) { 6288 return NO_INIT; 6289 } 6290 6291 sp<ThreadBase> thread = mThread.promote(); 6292 if (thread == 0) { 6293 return DEAD_OBJECT; 6294 } 6295 6296 // TODO: handle configuration of effects replacing track process 6297 if (thread->channelCount() == 1) { 6298 channels = AUDIO_CHANNEL_OUT_MONO; 6299 } else { 6300 channels = AUDIO_CHANNEL_OUT_STEREO; 6301 } 6302 6303 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6304 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6305 } else { 6306 mConfig.inputCfg.channels = channels; 6307 } 6308 mConfig.outputCfg.channels = channels; 6309 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6310 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6311 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6312 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6313 mConfig.inputCfg.bufferProvider.cookie = NULL; 6314 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6315 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6316 mConfig.outputCfg.bufferProvider.cookie = NULL; 6317 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6318 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6319 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6320 // Insert effect: 6321 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6322 // always overwrites output buffer: input buffer == output buffer 6323 // - in other sessions: 6324 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6325 // other effect: overwrites output buffer: input buffer == output buffer 6326 // Auxiliary effect: 6327 // accumulates in output buffer: input buffer != output buffer 6328 // Therefore: accumulate <=> input buffer != output buffer 6329 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6330 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6331 } else { 6332 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6333 } 6334 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6335 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6336 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6337 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6338 6339 ALOGV("configure() %p thread %p buffer %p framecount %d", 6340 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6341 6342 status_t cmdStatus; 6343 uint32_t size = sizeof(int); 6344 status_t status = (*mEffectInterface)->command(mEffectInterface, 6345 EFFECT_CMD_CONFIGURE, 6346 sizeof(effect_config_t), 6347 &mConfig, 6348 &size, 6349 &cmdStatus); 6350 if (status == 0) { 6351 status = cmdStatus; 6352 } 6353 6354 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6355 (1000 * mConfig.outputCfg.buffer.frameCount); 6356 6357 return status; 6358} 6359 6360status_t AudioFlinger::EffectModule::init() 6361{ 6362 Mutex::Autolock _l(mLock); 6363 if (mEffectInterface == NULL) { 6364 return NO_INIT; 6365 } 6366 status_t cmdStatus; 6367 uint32_t size = sizeof(status_t); 6368 status_t status = (*mEffectInterface)->command(mEffectInterface, 6369 EFFECT_CMD_INIT, 6370 0, 6371 NULL, 6372 &size, 6373 &cmdStatus); 6374 if (status == 0) { 6375 status = cmdStatus; 6376 } 6377 return status; 6378} 6379 6380status_t AudioFlinger::EffectModule::start() 6381{ 6382 Mutex::Autolock _l(mLock); 6383 return start_l(); 6384} 6385 6386status_t AudioFlinger::EffectModule::start_l() 6387{ 6388 if (mEffectInterface == NULL) { 6389 return NO_INIT; 6390 } 6391 status_t cmdStatus; 6392 uint32_t size = sizeof(status_t); 6393 status_t status = (*mEffectInterface)->command(mEffectInterface, 6394 EFFECT_CMD_ENABLE, 6395 0, 6396 NULL, 6397 &size, 6398 &cmdStatus); 6399 if (status == 0) { 6400 status = cmdStatus; 6401 } 6402 if (status == 0 && 6403 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6404 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6405 sp<ThreadBase> thread = mThread.promote(); 6406 if (thread != 0) { 6407 audio_stream_t *stream = thread->stream(); 6408 if (stream != NULL) { 6409 stream->add_audio_effect(stream, mEffectInterface); 6410 } 6411 } 6412 } 6413 return status; 6414} 6415 6416status_t AudioFlinger::EffectModule::stop() 6417{ 6418 Mutex::Autolock _l(mLock); 6419 return stop_l(); 6420} 6421 6422status_t AudioFlinger::EffectModule::stop_l() 6423{ 6424 if (mEffectInterface == NULL) { 6425 return NO_INIT; 6426 } 6427 status_t cmdStatus; 6428 uint32_t size = sizeof(status_t); 6429 status_t status = (*mEffectInterface)->command(mEffectInterface, 6430 EFFECT_CMD_DISABLE, 6431 0, 6432 NULL, 6433 &size, 6434 &cmdStatus); 6435 if (status == 0) { 6436 status = cmdStatus; 6437 } 6438 if (status == 0 && 6439 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6440 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6441 sp<ThreadBase> thread = mThread.promote(); 6442 if (thread != 0) { 6443 audio_stream_t *stream = thread->stream(); 6444 if (stream != NULL) { 6445 stream->remove_audio_effect(stream, mEffectInterface); 6446 } 6447 } 6448 } 6449 return status; 6450} 6451 6452status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6453 uint32_t cmdSize, 6454 void *pCmdData, 6455 uint32_t *replySize, 6456 void *pReplyData) 6457{ 6458 Mutex::Autolock _l(mLock); 6459// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6460 6461 if (mState == DESTROYED || mEffectInterface == NULL) { 6462 return NO_INIT; 6463 } 6464 status_t status = (*mEffectInterface)->command(mEffectInterface, 6465 cmdCode, 6466 cmdSize, 6467 pCmdData, 6468 replySize, 6469 pReplyData); 6470 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6471 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6472 for (size_t i = 1; i < mHandles.size(); i++) { 6473 sp<EffectHandle> h = mHandles[i].promote(); 6474 if (h != 0) { 6475 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6476 } 6477 } 6478 } 6479 return status; 6480} 6481 6482status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6483{ 6484 6485 Mutex::Autolock _l(mLock); 6486 ALOGV("setEnabled %p enabled %d", this, enabled); 6487 6488 if (enabled != isEnabled()) { 6489 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6490 if (enabled && status != NO_ERROR) { 6491 return status; 6492 } 6493 6494 switch (mState) { 6495 // going from disabled to enabled 6496 case IDLE: 6497 mState = STARTING; 6498 break; 6499 case STOPPED: 6500 mState = RESTART; 6501 break; 6502 case STOPPING: 6503 mState = ACTIVE; 6504 break; 6505 6506 // going from enabled to disabled 6507 case RESTART: 6508 mState = STOPPED; 6509 break; 6510 case STARTING: 6511 mState = IDLE; 6512 break; 6513 case ACTIVE: 6514 mState = STOPPING; 6515 break; 6516 case DESTROYED: 6517 return NO_ERROR; // simply ignore as we are being destroyed 6518 } 6519 for (size_t i = 1; i < mHandles.size(); i++) { 6520 sp<EffectHandle> h = mHandles[i].promote(); 6521 if (h != 0) { 6522 h->setEnabled(enabled); 6523 } 6524 } 6525 } 6526 return NO_ERROR; 6527} 6528 6529bool AudioFlinger::EffectModule::isEnabled() 6530{ 6531 switch (mState) { 6532 case RESTART: 6533 case STARTING: 6534 case ACTIVE: 6535 return true; 6536 case IDLE: 6537 case STOPPING: 6538 case STOPPED: 6539 case DESTROYED: 6540 default: 6541 return false; 6542 } 6543} 6544 6545bool AudioFlinger::EffectModule::isProcessEnabled() 6546{ 6547 switch (mState) { 6548 case RESTART: 6549 case ACTIVE: 6550 case STOPPING: 6551 case STOPPED: 6552 return true; 6553 case IDLE: 6554 case STARTING: 6555 case DESTROYED: 6556 default: 6557 return false; 6558 } 6559} 6560 6561status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6562{ 6563 Mutex::Autolock _l(mLock); 6564 status_t status = NO_ERROR; 6565 6566 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6567 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6568 if (isProcessEnabled() && 6569 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6570 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6571 status_t cmdStatus; 6572 uint32_t volume[2]; 6573 uint32_t *pVolume = NULL; 6574 uint32_t size = sizeof(volume); 6575 volume[0] = *left; 6576 volume[1] = *right; 6577 if (controller) { 6578 pVolume = volume; 6579 } 6580 status = (*mEffectInterface)->command(mEffectInterface, 6581 EFFECT_CMD_SET_VOLUME, 6582 size, 6583 volume, 6584 &size, 6585 pVolume); 6586 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6587 *left = volume[0]; 6588 *right = volume[1]; 6589 } 6590 } 6591 return status; 6592} 6593 6594status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6595{ 6596 Mutex::Autolock _l(mLock); 6597 status_t status = NO_ERROR; 6598 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6599 // audio pre processing modules on RecordThread can receive both output and 6600 // input device indication in the same call 6601 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6602 if (dev) { 6603 status_t cmdStatus; 6604 uint32_t size = sizeof(status_t); 6605 6606 status = (*mEffectInterface)->command(mEffectInterface, 6607 EFFECT_CMD_SET_DEVICE, 6608 sizeof(uint32_t), 6609 &dev, 6610 &size, 6611 &cmdStatus); 6612 if (status == NO_ERROR) { 6613 status = cmdStatus; 6614 } 6615 } 6616 dev = device & AUDIO_DEVICE_IN_ALL; 6617 if (dev) { 6618 status_t cmdStatus; 6619 uint32_t size = sizeof(status_t); 6620 6621 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6622 EFFECT_CMD_SET_INPUT_DEVICE, 6623 sizeof(uint32_t), 6624 &dev, 6625 &size, 6626 &cmdStatus); 6627 if (status2 == NO_ERROR) { 6628 status2 = cmdStatus; 6629 } 6630 if (status == NO_ERROR) { 6631 status = status2; 6632 } 6633 } 6634 } 6635 return status; 6636} 6637 6638status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6639{ 6640 Mutex::Autolock _l(mLock); 6641 status_t status = NO_ERROR; 6642 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6643 status_t cmdStatus; 6644 uint32_t size = sizeof(status_t); 6645 status = (*mEffectInterface)->command(mEffectInterface, 6646 EFFECT_CMD_SET_AUDIO_MODE, 6647 sizeof(int), 6648 &mode, 6649 &size, 6650 &cmdStatus); 6651 if (status == NO_ERROR) { 6652 status = cmdStatus; 6653 } 6654 } 6655 return status; 6656} 6657 6658void AudioFlinger::EffectModule::setSuspended(bool suspended) 6659{ 6660 Mutex::Autolock _l(mLock); 6661 mSuspended = suspended; 6662} 6663bool AudioFlinger::EffectModule::suspended() 6664{ 6665 Mutex::Autolock _l(mLock); 6666 return mSuspended; 6667} 6668 6669status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6670{ 6671 const size_t SIZE = 256; 6672 char buffer[SIZE]; 6673 String8 result; 6674 6675 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6676 result.append(buffer); 6677 6678 bool locked = tryLock(mLock); 6679 // failed to lock - AudioFlinger is probably deadlocked 6680 if (!locked) { 6681 result.append("\t\tCould not lock Fx mutex:\n"); 6682 } 6683 6684 result.append("\t\tSession Status State Engine:\n"); 6685 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6686 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6687 result.append(buffer); 6688 6689 result.append("\t\tDescriptor:\n"); 6690 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6691 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6692 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6693 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6694 result.append(buffer); 6695 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6696 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6697 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6698 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6699 result.append(buffer); 6700 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6701 mDescriptor.apiVersion, 6702 mDescriptor.flags); 6703 result.append(buffer); 6704 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6705 mDescriptor.name); 6706 result.append(buffer); 6707 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6708 mDescriptor.implementor); 6709 result.append(buffer); 6710 6711 result.append("\t\t- Input configuration:\n"); 6712 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6713 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6714 (uint32_t)mConfig.inputCfg.buffer.raw, 6715 mConfig.inputCfg.buffer.frameCount, 6716 mConfig.inputCfg.samplingRate, 6717 mConfig.inputCfg.channels, 6718 mConfig.inputCfg.format); 6719 result.append(buffer); 6720 6721 result.append("\t\t- Output configuration:\n"); 6722 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6723 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6724 (uint32_t)mConfig.outputCfg.buffer.raw, 6725 mConfig.outputCfg.buffer.frameCount, 6726 mConfig.outputCfg.samplingRate, 6727 mConfig.outputCfg.channels, 6728 mConfig.outputCfg.format); 6729 result.append(buffer); 6730 6731 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6732 result.append(buffer); 6733 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6734 for (size_t i = 0; i < mHandles.size(); ++i) { 6735 sp<EffectHandle> handle = mHandles[i].promote(); 6736 if (handle != 0) { 6737 handle->dump(buffer, SIZE); 6738 result.append(buffer); 6739 } 6740 } 6741 6742 result.append("\n"); 6743 6744 write(fd, result.string(), result.length()); 6745 6746 if (locked) { 6747 mLock.unlock(); 6748 } 6749 6750 return NO_ERROR; 6751} 6752 6753// ---------------------------------------------------------------------------- 6754// EffectHandle implementation 6755// ---------------------------------------------------------------------------- 6756 6757#undef LOG_TAG 6758#define LOG_TAG "AudioFlinger::EffectHandle" 6759 6760AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6761 const sp<AudioFlinger::Client>& client, 6762 const sp<IEffectClient>& effectClient, 6763 int32_t priority) 6764 : BnEffect(), 6765 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6766 mPriority(priority), mHasControl(false), mEnabled(false) 6767{ 6768 ALOGV("constructor %p", this); 6769 6770 if (client == 0) { 6771 return; 6772 } 6773 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6774 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6775 if (mCblkMemory != 0) { 6776 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6777 6778 if (mCblk) { 6779 new(mCblk) effect_param_cblk_t(); 6780 mBuffer = (uint8_t *)mCblk + bufOffset; 6781 } 6782 } else { 6783 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6784 return; 6785 } 6786} 6787 6788AudioFlinger::EffectHandle::~EffectHandle() 6789{ 6790 ALOGV("Destructor %p", this); 6791 disconnect(false); 6792 ALOGV("Destructor DONE %p", this); 6793} 6794 6795status_t AudioFlinger::EffectHandle::enable() 6796{ 6797 ALOGV("enable %p", this); 6798 if (!mHasControl) return INVALID_OPERATION; 6799 if (mEffect == 0) return DEAD_OBJECT; 6800 6801 if (mEnabled) { 6802 return NO_ERROR; 6803 } 6804 6805 mEnabled = true; 6806 6807 sp<ThreadBase> thread = mEffect->thread().promote(); 6808 if (thread != 0) { 6809 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6810 } 6811 6812 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6813 if (mEffect->suspended()) { 6814 return NO_ERROR; 6815 } 6816 6817 status_t status = mEffect->setEnabled(true); 6818 if (status != NO_ERROR) { 6819 if (thread != 0) { 6820 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6821 } 6822 mEnabled = false; 6823 } 6824 return status; 6825} 6826 6827status_t AudioFlinger::EffectHandle::disable() 6828{ 6829 ALOGV("disable %p", this); 6830 if (!mHasControl) return INVALID_OPERATION; 6831 if (mEffect == 0) return DEAD_OBJECT; 6832 6833 if (!mEnabled) { 6834 return NO_ERROR; 6835 } 6836 mEnabled = false; 6837 6838 if (mEffect->suspended()) { 6839 return NO_ERROR; 6840 } 6841 6842 status_t status = mEffect->setEnabled(false); 6843 6844 sp<ThreadBase> thread = mEffect->thread().promote(); 6845 if (thread != 0) { 6846 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6847 } 6848 6849 return status; 6850} 6851 6852void AudioFlinger::EffectHandle::disconnect() 6853{ 6854 disconnect(true); 6855} 6856 6857void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6858{ 6859 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6860 if (mEffect == 0) { 6861 return; 6862 } 6863 mEffect->disconnect(this, unpiniflast); 6864 6865 if (mHasControl && mEnabled) { 6866 sp<ThreadBase> thread = mEffect->thread().promote(); 6867 if (thread != 0) { 6868 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6869 } 6870 } 6871 6872 // release sp on module => module destructor can be called now 6873 mEffect.clear(); 6874 if (mClient != 0) { 6875 if (mCblk) { 6876 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6877 } 6878 mCblkMemory.clear(); // and free the shared memory 6879 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6880 mClient.clear(); 6881 } 6882} 6883 6884status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6885 uint32_t cmdSize, 6886 void *pCmdData, 6887 uint32_t *replySize, 6888 void *pReplyData) 6889{ 6890// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6891// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6892 6893 // only get parameter command is permitted for applications not controlling the effect 6894 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6895 return INVALID_OPERATION; 6896 } 6897 if (mEffect == 0) return DEAD_OBJECT; 6898 if (mClient == 0) return INVALID_OPERATION; 6899 6900 // handle commands that are not forwarded transparently to effect engine 6901 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6902 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6903 // no risk to block the whole media server process or mixer threads is we are stuck here 6904 Mutex::Autolock _l(mCblk->lock); 6905 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6906 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6907 mCblk->serverIndex = 0; 6908 mCblk->clientIndex = 0; 6909 return BAD_VALUE; 6910 } 6911 status_t status = NO_ERROR; 6912 while (mCblk->serverIndex < mCblk->clientIndex) { 6913 int reply; 6914 uint32_t rsize = sizeof(int); 6915 int *p = (int *)(mBuffer + mCblk->serverIndex); 6916 int size = *p++; 6917 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6918 LOGW("command(): invalid parameter block size"); 6919 break; 6920 } 6921 effect_param_t *param = (effect_param_t *)p; 6922 if (param->psize == 0 || param->vsize == 0) { 6923 LOGW("command(): null parameter or value size"); 6924 mCblk->serverIndex += size; 6925 continue; 6926 } 6927 uint32_t psize = sizeof(effect_param_t) + 6928 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6929 param->vsize; 6930 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6931 psize, 6932 p, 6933 &rsize, 6934 &reply); 6935 // stop at first error encountered 6936 if (ret != NO_ERROR) { 6937 status = ret; 6938 *(int *)pReplyData = reply; 6939 break; 6940 } else if (reply != NO_ERROR) { 6941 *(int *)pReplyData = reply; 6942 break; 6943 } 6944 mCblk->serverIndex += size; 6945 } 6946 mCblk->serverIndex = 0; 6947 mCblk->clientIndex = 0; 6948 return status; 6949 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6950 *(int *)pReplyData = NO_ERROR; 6951 return enable(); 6952 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6953 *(int *)pReplyData = NO_ERROR; 6954 return disable(); 6955 } 6956 6957 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6958} 6959 6960sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6961 return mCblkMemory; 6962} 6963 6964void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 6965{ 6966 ALOGV("setControl %p control %d", this, hasControl); 6967 6968 mHasControl = hasControl; 6969 mEnabled = enabled; 6970 6971 if (signal && mEffectClient != 0) { 6972 mEffectClient->controlStatusChanged(hasControl); 6973 } 6974} 6975 6976void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6977 uint32_t cmdSize, 6978 void *pCmdData, 6979 uint32_t replySize, 6980 void *pReplyData) 6981{ 6982 if (mEffectClient != 0) { 6983 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6984 } 6985} 6986 6987 6988 6989void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6990{ 6991 if (mEffectClient != 0) { 6992 mEffectClient->enableStatusChanged(enabled); 6993 } 6994} 6995 6996status_t AudioFlinger::EffectHandle::onTransact( 6997 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6998{ 6999 return BnEffect::onTransact(code, data, reply, flags); 7000} 7001 7002 7003void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7004{ 7005 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7006 7007 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7008 (mClient == NULL) ? getpid() : mClient->pid(), 7009 mPriority, 7010 mHasControl, 7011 !locked, 7012 mCblk ? mCblk->clientIndex : 0, 7013 mCblk ? mCblk->serverIndex : 0 7014 ); 7015 7016 if (locked) { 7017 mCblk->lock.unlock(); 7018 } 7019} 7020 7021#undef LOG_TAG 7022#define LOG_TAG "AudioFlinger::EffectChain" 7023 7024AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7025 int sessionId) 7026 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), 7027 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7028 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7029{ 7030 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7031} 7032 7033AudioFlinger::EffectChain::~EffectChain() 7034{ 7035 if (mOwnInBuffer) { 7036 delete mInBuffer; 7037 } 7038 7039} 7040 7041// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7042sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7043{ 7044 sp<EffectModule> effect; 7045 size_t size = mEffects.size(); 7046 7047 for (size_t i = 0; i < size; i++) { 7048 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7049 effect = mEffects[i]; 7050 break; 7051 } 7052 } 7053 return effect; 7054} 7055 7056// getEffectFromId_l() must be called with ThreadBase::mLock held 7057sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7058{ 7059 sp<EffectModule> effect; 7060 size_t size = mEffects.size(); 7061 7062 for (size_t i = 0; i < size; i++) { 7063 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7064 if (id == 0 || mEffects[i]->id() == id) { 7065 effect = mEffects[i]; 7066 break; 7067 } 7068 } 7069 return effect; 7070} 7071 7072// getEffectFromType_l() must be called with ThreadBase::mLock held 7073sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7074 const effect_uuid_t *type) 7075{ 7076 sp<EffectModule> effect; 7077 size_t size = mEffects.size(); 7078 7079 for (size_t i = 0; i < size; i++) { 7080 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7081 effect = mEffects[i]; 7082 break; 7083 } 7084 } 7085 return effect; 7086} 7087 7088// Must be called with EffectChain::mLock locked 7089void AudioFlinger::EffectChain::process_l() 7090{ 7091 sp<ThreadBase> thread = mThread.promote(); 7092 if (thread == 0) { 7093 LOGW("process_l(): cannot promote mixer thread"); 7094 return; 7095 } 7096 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7097 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7098 bool tracksOnSession = false; 7099 if (!isGlobalSession) { 7100 tracksOnSession = (trackCnt() != 0); 7101 } 7102 7103 // if no track is active, input buffer must be cleared here as the mixer process 7104 // will not do it 7105 if (tracksOnSession && 7106 activeTrackCnt() == 0) { 7107 size_t numSamples = thread->frameCount() * thread->channelCount(); 7108 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7109 } 7110 7111 size_t size = mEffects.size(); 7112 // do not process effect if no track is present in same audio session 7113 if (isGlobalSession || tracksOnSession) { 7114 for (size_t i = 0; i < size; i++) { 7115 mEffects[i]->process(); 7116 } 7117 } 7118 for (size_t i = 0; i < size; i++) { 7119 mEffects[i]->updateState(); 7120 } 7121} 7122 7123// addEffect_l() must be called with PlaybackThread::mLock held 7124status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7125{ 7126 effect_descriptor_t desc = effect->desc(); 7127 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7128 7129 Mutex::Autolock _l(mLock); 7130 effect->setChain(this); 7131 sp<ThreadBase> thread = mThread.promote(); 7132 if (thread == 0) { 7133 return NO_INIT; 7134 } 7135 effect->setThread(thread); 7136 7137 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7138 // Auxiliary effects are inserted at the beginning of mEffects vector as 7139 // they are processed first and accumulated in chain input buffer 7140 mEffects.insertAt(effect, 0); 7141 7142 // the input buffer for auxiliary effect contains mono samples in 7143 // 32 bit format. This is to avoid saturation in AudoMixer 7144 // accumulation stage. Saturation is done in EffectModule::process() before 7145 // calling the process in effect engine 7146 size_t numSamples = thread->frameCount(); 7147 int32_t *buffer = new int32_t[numSamples]; 7148 memset(buffer, 0, numSamples * sizeof(int32_t)); 7149 effect->setInBuffer((int16_t *)buffer); 7150 // auxiliary effects output samples to chain input buffer for further processing 7151 // by insert effects 7152 effect->setOutBuffer(mInBuffer); 7153 } else { 7154 // Insert effects are inserted at the end of mEffects vector as they are processed 7155 // after track and auxiliary effects. 7156 // Insert effect order as a function of indicated preference: 7157 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7158 // another effect is present 7159 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7160 // last effect claiming first position 7161 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7162 // first effect claiming last position 7163 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7164 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7165 // already present 7166 7167 int size = (int)mEffects.size(); 7168 int idx_insert = size; 7169 int idx_insert_first = -1; 7170 int idx_insert_last = -1; 7171 7172 for (int i = 0; i < size; i++) { 7173 effect_descriptor_t d = mEffects[i]->desc(); 7174 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7175 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7176 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7177 // check invalid effect chaining combinations 7178 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7179 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7180 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7181 return INVALID_OPERATION; 7182 } 7183 // remember position of first insert effect and by default 7184 // select this as insert position for new effect 7185 if (idx_insert == size) { 7186 idx_insert = i; 7187 } 7188 // remember position of last insert effect claiming 7189 // first position 7190 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7191 idx_insert_first = i; 7192 } 7193 // remember position of first insert effect claiming 7194 // last position 7195 if (iPref == EFFECT_FLAG_INSERT_LAST && 7196 idx_insert_last == -1) { 7197 idx_insert_last = i; 7198 } 7199 } 7200 } 7201 7202 // modify idx_insert from first position if needed 7203 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7204 if (idx_insert_last != -1) { 7205 idx_insert = idx_insert_last; 7206 } else { 7207 idx_insert = size; 7208 } 7209 } else { 7210 if (idx_insert_first != -1) { 7211 idx_insert = idx_insert_first + 1; 7212 } 7213 } 7214 7215 // always read samples from chain input buffer 7216 effect->setInBuffer(mInBuffer); 7217 7218 // if last effect in the chain, output samples to chain 7219 // output buffer, otherwise to chain input buffer 7220 if (idx_insert == size) { 7221 if (idx_insert != 0) { 7222 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7223 mEffects[idx_insert-1]->configure(); 7224 } 7225 effect->setOutBuffer(mOutBuffer); 7226 } else { 7227 effect->setOutBuffer(mInBuffer); 7228 } 7229 mEffects.insertAt(effect, idx_insert); 7230 7231 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7232 } 7233 effect->configure(); 7234 return NO_ERROR; 7235} 7236 7237// removeEffect_l() must be called with PlaybackThread::mLock held 7238size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7239{ 7240 Mutex::Autolock _l(mLock); 7241 int size = (int)mEffects.size(); 7242 int i; 7243 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7244 7245 for (i = 0; i < size; i++) { 7246 if (effect == mEffects[i]) { 7247 // calling stop here will remove pre-processing effect from the audio HAL. 7248 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7249 // the middle of a read from audio HAL 7250 if (mEffects[i]->state() == EffectModule::ACTIVE || 7251 mEffects[i]->state() == EffectModule::STOPPING) { 7252 mEffects[i]->stop(); 7253 } 7254 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7255 delete[] effect->inBuffer(); 7256 } else { 7257 if (i == size - 1 && i != 0) { 7258 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7259 mEffects[i - 1]->configure(); 7260 } 7261 } 7262 mEffects.removeAt(i); 7263 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7264 break; 7265 } 7266 } 7267 7268 return mEffects.size(); 7269} 7270 7271// setDevice_l() must be called with PlaybackThread::mLock held 7272void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7273{ 7274 size_t size = mEffects.size(); 7275 for (size_t i = 0; i < size; i++) { 7276 mEffects[i]->setDevice(device); 7277 } 7278} 7279 7280// setMode_l() must be called with PlaybackThread::mLock held 7281void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 7282{ 7283 size_t size = mEffects.size(); 7284 for (size_t i = 0; i < size; i++) { 7285 mEffects[i]->setMode(mode); 7286 } 7287} 7288 7289// setVolume_l() must be called with PlaybackThread::mLock held 7290bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7291{ 7292 uint32_t newLeft = *left; 7293 uint32_t newRight = *right; 7294 bool hasControl = false; 7295 int ctrlIdx = -1; 7296 size_t size = mEffects.size(); 7297 7298 // first update volume controller 7299 for (size_t i = size; i > 0; i--) { 7300 if (mEffects[i - 1]->isProcessEnabled() && 7301 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7302 ctrlIdx = i - 1; 7303 hasControl = true; 7304 break; 7305 } 7306 } 7307 7308 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7309 if (hasControl) { 7310 *left = mNewLeftVolume; 7311 *right = mNewRightVolume; 7312 } 7313 return hasControl; 7314 } 7315 7316 mVolumeCtrlIdx = ctrlIdx; 7317 mLeftVolume = newLeft; 7318 mRightVolume = newRight; 7319 7320 // second get volume update from volume controller 7321 if (ctrlIdx >= 0) { 7322 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7323 mNewLeftVolume = newLeft; 7324 mNewRightVolume = newRight; 7325 } 7326 // then indicate volume to all other effects in chain. 7327 // Pass altered volume to effects before volume controller 7328 // and requested volume to effects after controller 7329 uint32_t lVol = newLeft; 7330 uint32_t rVol = newRight; 7331 7332 for (size_t i = 0; i < size; i++) { 7333 if ((int)i == ctrlIdx) continue; 7334 // this also works for ctrlIdx == -1 when there is no volume controller 7335 if ((int)i > ctrlIdx) { 7336 lVol = *left; 7337 rVol = *right; 7338 } 7339 mEffects[i]->setVolume(&lVol, &rVol, false); 7340 } 7341 *left = newLeft; 7342 *right = newRight; 7343 7344 return hasControl; 7345} 7346 7347status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7348{ 7349 const size_t SIZE = 256; 7350 char buffer[SIZE]; 7351 String8 result; 7352 7353 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7354 result.append(buffer); 7355 7356 bool locked = tryLock(mLock); 7357 // failed to lock - AudioFlinger is probably deadlocked 7358 if (!locked) { 7359 result.append("\tCould not lock mutex:\n"); 7360 } 7361 7362 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7363 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7364 mEffects.size(), 7365 (uint32_t)mInBuffer, 7366 (uint32_t)mOutBuffer, 7367 mActiveTrackCnt); 7368 result.append(buffer); 7369 write(fd, result.string(), result.size()); 7370 7371 for (size_t i = 0; i < mEffects.size(); ++i) { 7372 sp<EffectModule> effect = mEffects[i]; 7373 if (effect != 0) { 7374 effect->dump(fd, args); 7375 } 7376 } 7377 7378 if (locked) { 7379 mLock.unlock(); 7380 } 7381 7382 return NO_ERROR; 7383} 7384 7385// must be called with ThreadBase::mLock held 7386void AudioFlinger::EffectChain::setEffectSuspended_l( 7387 const effect_uuid_t *type, bool suspend) 7388{ 7389 sp<SuspendedEffectDesc> desc; 7390 // use effect type UUID timelow as key as there is no real risk of identical 7391 // timeLow fields among effect type UUIDs. 7392 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7393 if (suspend) { 7394 if (index >= 0) { 7395 desc = mSuspendedEffects.valueAt(index); 7396 } else { 7397 desc = new SuspendedEffectDesc(); 7398 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7399 mSuspendedEffects.add(type->timeLow, desc); 7400 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7401 } 7402 if (desc->mRefCount++ == 0) { 7403 sp<EffectModule> effect = getEffectIfEnabled(type); 7404 if (effect != 0) { 7405 desc->mEffect = effect; 7406 effect->setSuspended(true); 7407 effect->setEnabled(false); 7408 } 7409 } 7410 } else { 7411 if (index < 0) { 7412 return; 7413 } 7414 desc = mSuspendedEffects.valueAt(index); 7415 if (desc->mRefCount <= 0) { 7416 LOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7417 desc->mRefCount = 1; 7418 } 7419 if (--desc->mRefCount == 0) { 7420 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7421 if (desc->mEffect != 0) { 7422 sp<EffectModule> effect = desc->mEffect.promote(); 7423 if (effect != 0) { 7424 effect->setSuspended(false); 7425 sp<EffectHandle> handle = effect->controlHandle(); 7426 if (handle != 0) { 7427 effect->setEnabled(handle->enabled()); 7428 } 7429 } 7430 desc->mEffect.clear(); 7431 } 7432 mSuspendedEffects.removeItemsAt(index); 7433 } 7434 } 7435} 7436 7437// must be called with ThreadBase::mLock held 7438void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7439{ 7440 sp<SuspendedEffectDesc> desc; 7441 7442 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7443 if (suspend) { 7444 if (index >= 0) { 7445 desc = mSuspendedEffects.valueAt(index); 7446 } else { 7447 desc = new SuspendedEffectDesc(); 7448 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7449 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7450 } 7451 if (desc->mRefCount++ == 0) { 7452 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7453 for (size_t i = 0; i < effects.size(); i++) { 7454 setEffectSuspended_l(&effects[i]->desc().type, true); 7455 } 7456 } 7457 } else { 7458 if (index < 0) { 7459 return; 7460 } 7461 desc = mSuspendedEffects.valueAt(index); 7462 if (desc->mRefCount <= 0) { 7463 LOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7464 desc->mRefCount = 1; 7465 } 7466 if (--desc->mRefCount == 0) { 7467 Vector<const effect_uuid_t *> types; 7468 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7469 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7470 continue; 7471 } 7472 types.add(&mSuspendedEffects.valueAt(i)->mType); 7473 } 7474 for (size_t i = 0; i < types.size(); i++) { 7475 setEffectSuspended_l(types[i], false); 7476 } 7477 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7478 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7479 } 7480 } 7481} 7482 7483 7484// The volume effect is used for automated tests only 7485#ifndef OPENSL_ES_H_ 7486static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7487 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7488const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7489#endif //OPENSL_ES_H_ 7490 7491bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7492{ 7493 // auxiliary effects and visualizer are never suspended on output mix 7494 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7495 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7496 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7497 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7498 return false; 7499 } 7500 return true; 7501} 7502 7503Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7504{ 7505 Vector< sp<EffectModule> > effects; 7506 for (size_t i = 0; i < mEffects.size(); i++) { 7507 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7508 continue; 7509 } 7510 effects.add(mEffects[i]); 7511 } 7512 return effects; 7513} 7514 7515sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7516 const effect_uuid_t *type) 7517{ 7518 sp<EffectModule> effect; 7519 effect = getEffectFromType_l(type); 7520 if (effect != 0 && !effect->isEnabled()) { 7521 effect.clear(); 7522 } 7523 return effect; 7524} 7525 7526void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7527 bool enabled) 7528{ 7529 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7530 if (enabled) { 7531 if (index < 0) { 7532 // if the effect is not suspend check if all effects are suspended 7533 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7534 if (index < 0) { 7535 return; 7536 } 7537 if (!isEffectEligibleForSuspend(effect->desc())) { 7538 return; 7539 } 7540 setEffectSuspended_l(&effect->desc().type, enabled); 7541 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7542 if (index < 0) { 7543 LOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7544 return; 7545 } 7546 } 7547 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7548 effect->desc().type.timeLow); 7549 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7550 // if effect is requested to suspended but was not yet enabled, supend it now. 7551 if (desc->mEffect == 0) { 7552 desc->mEffect = effect; 7553 effect->setEnabled(false); 7554 effect->setSuspended(true); 7555 } 7556 } else { 7557 if (index < 0) { 7558 return; 7559 } 7560 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7561 effect->desc().type.timeLow); 7562 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7563 desc->mEffect.clear(); 7564 effect->setSuspended(false); 7565 } 7566} 7567 7568#undef LOG_TAG 7569#define LOG_TAG "AudioFlinger" 7570 7571// ---------------------------------------------------------------------------- 7572 7573status_t AudioFlinger::onTransact( 7574 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7575{ 7576 return BnAudioFlinger::onTransact(code, data, reply, flags); 7577} 7578 7579}; // namespace android 7580