AudioFlinger.cpp revision 000f0e39b4d0c88441297a05ab5f8da6832c1640
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 mNotificationClients.removeItem(pid); 1033 1034 ALOGV("%d died, releasing its sessions", pid); 1035 size_t num = mAudioSessionRefs.size(); 1036 bool removed = false; 1037 for (size_t i = 0; i< num; ) { 1038 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1039 ALOGV(" pid %d @ %d", ref->pid, i); 1040 if (ref->pid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1042 mAudioSessionRefs.removeAt(i); 1043 delete ref; 1044 removed = true; 1045 num--; 1046 } else { 1047 i++; 1048 } 1049 } 1050 if (removed) { 1051 purgeStaleEffects_l(); 1052 } 1053} 1054 1055// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1056void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1057{ 1058 size_t size = mNotificationClients.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1061 param2); 1062 } 1063} 1064 1065// removeClient_l() must be called with AudioFlinger::mLock held 1066void AudioFlinger::removeClient_l(pid_t pid) 1067{ 1068 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1069 mClients.removeItem(pid); 1070} 1071 1072 1073// ---------------------------------------------------------------------------- 1074 1075AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1076 uint32_t device, type_t type) 1077 : Thread(false), 1078 mType(type), 1079 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1080 // mChannelMask 1081 mChannelCount(0), 1082 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1083 mParamStatus(NO_ERROR), 1084 mStandby(false), mId(id), 1085 mDevice(device), 1086 mDeathRecipient(new PMDeathRecipient(this)) 1087{ 1088} 1089 1090AudioFlinger::ThreadBase::~ThreadBase() 1091{ 1092 mParamCond.broadcast(); 1093 // do not lock the mutex in destructor 1094 releaseWakeLock_l(); 1095 if (mPowerManager != 0) { 1096 sp<IBinder> binder = mPowerManager->asBinder(); 1097 binder->unlinkToDeath(mDeathRecipient); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::exit() 1102{ 1103 ALOGV("ThreadBase::exit"); 1104 { 1105 // This lock prevents the following race in thread (uniprocessor for illustration): 1106 // if (!exitPending()) { 1107 // // context switch from here to exit() 1108 // // exit() calls requestExit(), what exitPending() observes 1109 // // exit() calls signal(), which is dropped since no waiters 1110 // // context switch back from exit() to here 1111 // mWaitWorkCV.wait(...); 1112 // // now thread is hung 1113 // } 1114 AutoMutex lock(mLock); 1115 requestExit(); 1116 mWaitWorkCV.signal(); 1117 } 1118 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1119 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1120 requestExitAndWait(); 1121} 1122 1123status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1124{ 1125 status_t status; 1126 1127 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1128 Mutex::Autolock _l(mLock); 1129 1130 mNewParameters.add(keyValuePairs); 1131 mWaitWorkCV.signal(); 1132 // wait condition with timeout in case the thread loop has exited 1133 // before the request could be processed 1134 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1135 status = mParamStatus; 1136 mWaitWorkCV.signal(); 1137 } else { 1138 status = TIMED_OUT; 1139 } 1140 return status; 1141} 1142 1143void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1144{ 1145 Mutex::Autolock _l(mLock); 1146 sendConfigEvent_l(event, param); 1147} 1148 1149// sendConfigEvent_l() must be called with ThreadBase::mLock held 1150void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1151{ 1152 ConfigEvent configEvent; 1153 configEvent.mEvent = event; 1154 configEvent.mParam = param; 1155 mConfigEvents.add(configEvent); 1156 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1157 mWaitWorkCV.signal(); 1158} 1159 1160void AudioFlinger::ThreadBase::processConfigEvents() 1161{ 1162 mLock.lock(); 1163 while(!mConfigEvents.isEmpty()) { 1164 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1165 ConfigEvent configEvent = mConfigEvents[0]; 1166 mConfigEvents.removeAt(0); 1167 // release mLock before locking AudioFlinger mLock: lock order is always 1168 // AudioFlinger then ThreadBase to avoid cross deadlock 1169 mLock.unlock(); 1170 mAudioFlinger->mLock.lock(); 1171 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1172 mAudioFlinger->mLock.unlock(); 1173 mLock.lock(); 1174 } 1175 mLock.unlock(); 1176} 1177 1178status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1179{ 1180 const size_t SIZE = 256; 1181 char buffer[SIZE]; 1182 String8 result; 1183 1184 bool locked = tryLock(mLock); 1185 if (!locked) { 1186 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1187 write(fd, buffer, strlen(buffer)); 1188 } 1189 1190 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1203 result.append(buffer); 1204 1205 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1206 result.append(buffer); 1207 result.append(" Index Command"); 1208 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1209 snprintf(buffer, SIZE, "\n %02d ", i); 1210 result.append(buffer); 1211 result.append(mNewParameters[i]); 1212 } 1213 1214 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, " Index event param\n"); 1217 result.append(buffer); 1218 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1219 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1220 result.append(buffer); 1221 } 1222 result.append("\n"); 1223 1224 write(fd, result.string(), result.size()); 1225 1226 if (locked) { 1227 mLock.unlock(); 1228 } 1229 return NO_ERROR; 1230} 1231 1232status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1233{ 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1239 write(fd, buffer, strlen(buffer)); 1240 1241 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1242 sp<EffectChain> chain = mEffectChains[i]; 1243 if (chain != 0) { 1244 chain->dump(fd, args); 1245 } 1246 } 1247 return NO_ERROR; 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock() 1251{ 1252 Mutex::Autolock _l(mLock); 1253 acquireWakeLock_l(); 1254} 1255 1256void AudioFlinger::ThreadBase::acquireWakeLock_l() 1257{ 1258 if (mPowerManager == 0) { 1259 // use checkService() to avoid blocking if power service is not up yet 1260 sp<IBinder> binder = 1261 defaultServiceManager()->checkService(String16("power")); 1262 if (binder == 0) { 1263 ALOGW("Thread %s cannot connect to the power manager service", mName); 1264 } else { 1265 mPowerManager = interface_cast<IPowerManager>(binder); 1266 binder->linkToDeath(mDeathRecipient); 1267 } 1268 } 1269 if (mPowerManager != 0) { 1270 sp<IBinder> binder = new BBinder(); 1271 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1272 binder, 1273 String16(mName)); 1274 if (status == NO_ERROR) { 1275 mWakeLockToken = binder; 1276 } 1277 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock() 1282{ 1283 Mutex::Autolock _l(mLock); 1284 releaseWakeLock_l(); 1285} 1286 1287void AudioFlinger::ThreadBase::releaseWakeLock_l() 1288{ 1289 if (mWakeLockToken != 0) { 1290 ALOGV("releaseWakeLock_l() %s", mName); 1291 if (mPowerManager != 0) { 1292 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1293 } 1294 mWakeLockToken.clear(); 1295 } 1296} 1297 1298void AudioFlinger::ThreadBase::clearPowerManager() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 releaseWakeLock_l(); 1302 mPowerManager.clear(); 1303} 1304 1305void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1306{ 1307 sp<ThreadBase> thread = mThread.promote(); 1308 if (thread != 0) { 1309 thread->clearPowerManager(); 1310 } 1311 ALOGW("power manager service died !!!"); 1312} 1313 1314void AudioFlinger::ThreadBase::setEffectSuspended( 1315 const effect_uuid_t *type, bool suspend, int sessionId) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 setEffectSuspended_l(type, suspend, sessionId); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended_l( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 sp<EffectChain> chain = getEffectChain_l(sessionId); 1325 if (chain != 0) { 1326 if (type != NULL) { 1327 chain->setEffectSuspended_l(type, suspend); 1328 } else { 1329 chain->setEffectSuspendedAll_l(suspend); 1330 } 1331 } 1332 1333 updateSuspendedSessions_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1337{ 1338 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1339 if (index < 0) { 1340 return; 1341 } 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1344 mSuspendedSessions.editValueAt(index); 1345 1346 for (size_t i = 0; i < sessionEffects.size(); i++) { 1347 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1348 for (int j = 0; j < desc->mRefCount; j++) { 1349 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1350 chain->setEffectSuspendedAll_l(true); 1351 } else { 1352 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1353 desc->mType.timeLow); 1354 chain->setEffectSuspended_l(&desc->mType, true); 1355 } 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1361 bool suspend, 1362 int sessionId) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1365 1366 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1367 1368 if (suspend) { 1369 if (index >= 0) { 1370 sessionEffects = mSuspendedSessions.editValueAt(index); 1371 } else { 1372 mSuspendedSessions.add(sessionId, sessionEffects); 1373 } 1374 } else { 1375 if (index < 0) { 1376 return; 1377 } 1378 sessionEffects = mSuspendedSessions.editValueAt(index); 1379 } 1380 1381 1382 int key = EffectChain::kKeyForSuspendAll; 1383 if (type != NULL) { 1384 key = type->timeLow; 1385 } 1386 index = sessionEffects.indexOfKey(key); 1387 1388 sp <SuspendedSessionDesc> desc; 1389 if (suspend) { 1390 if (index >= 0) { 1391 desc = sessionEffects.valueAt(index); 1392 } else { 1393 desc = new SuspendedSessionDesc(); 1394 if (type != NULL) { 1395 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1396 } 1397 sessionEffects.add(key, desc); 1398 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1399 } 1400 desc->mRefCount++; 1401 } else { 1402 if (index < 0) { 1403 return; 1404 } 1405 desc = sessionEffects.valueAt(index); 1406 if (--desc->mRefCount == 0) { 1407 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1408 sessionEffects.removeItemsAt(index); 1409 if (sessionEffects.isEmpty()) { 1410 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1411 sessionId); 1412 mSuspendedSessions.removeItem(sessionId); 1413 } 1414 } 1415 } 1416 if (!sessionEffects.isEmpty()) { 1417 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1418 } 1419} 1420 1421void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1422 bool enabled, 1423 int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1430 bool enabled, 1431 int sessionId) 1432{ 1433 if (mType != RECORD) { 1434 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1435 // another session. This gives the priority to well behaved effect control panels 1436 // and applications not using global effects. 1437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1438 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1439 } 1440 } 1441 1442 sp<EffectChain> chain = getEffectChain_l(sessionId); 1443 if (chain != 0) { 1444 chain->checkSuspendOnEffectEnabled(effect, enabled); 1445 } 1446} 1447 1448// ---------------------------------------------------------------------------- 1449 1450AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1451 AudioStreamOut* output, 1452 audio_io_handle_t id, 1453 uint32_t device, 1454 type_t type) 1455 : ThreadBase(audioFlinger, id, device, type), 1456 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1457 // Assumes constructor is called by AudioFlinger with it's mLock held, 1458 // but it would be safer to explicitly pass initial masterMute as parameter 1459 mMasterMute(audioFlinger->masterMute_l()), 1460 // mStreamTypes[] initialized in constructor body 1461 mOutput(output), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterVolume as parameter 1464 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1465 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1466{ 1467 snprintf(mName, kNameLength, "AudioOut_%X", id); 1468 1469 readOutputParameters(); 1470 1471 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1472 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1473 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1474 stream = (audio_stream_type_t) (stream + 1)) { 1475 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1476 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1477 // initialized by stream_type_t default constructor 1478 // mStreamTypes[stream].valid = true; 1479 } 1480 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1481 // because mAudioFlinger doesn't have one to copy from 1482} 1483 1484AudioFlinger::PlaybackThread::~PlaybackThread() 1485{ 1486 delete [] mMixBuffer; 1487} 1488 1489status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1490{ 1491 dumpInternals(fd, args); 1492 dumpTracks(fd, args); 1493 dumpEffectChains(fd, args); 1494 return NO_ERROR; 1495} 1496 1497status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1498{ 1499 const size_t SIZE = 256; 1500 char buffer[SIZE]; 1501 String8 result; 1502 1503 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1504 result.append(buffer); 1505 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1506 for (size_t i = 0; i < mTracks.size(); ++i) { 1507 sp<Track> track = mTracks[i]; 1508 if (track != 0) { 1509 track->dump(buffer, SIZE); 1510 result.append(buffer); 1511 } 1512 } 1513 1514 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1515 result.append(buffer); 1516 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1517 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1518 sp<Track> track = mActiveTracks[i].promote(); 1519 if (track != 0) { 1520 track->dump(buffer, SIZE); 1521 result.append(buffer); 1522 } 1523 } 1524 write(fd, result.string(), result.size()); 1525 return NO_ERROR; 1526} 1527 1528status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1547 result.append(buffer); 1548 write(fd, result.string(), result.size()); 1549 1550 dumpBase(fd, args); 1551 1552 return NO_ERROR; 1553} 1554 1555// Thread virtuals 1556status_t AudioFlinger::PlaybackThread::readyToRun() 1557{ 1558 status_t status = initCheck(); 1559 if (status == NO_ERROR) { 1560 ALOGI("AudioFlinger's thread %p ready to run", this); 1561 } else { 1562 ALOGE("No working audio driver found."); 1563 } 1564 return status; 1565} 1566 1567void AudioFlinger::PlaybackThread::onFirstRef() 1568{ 1569 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1570} 1571 1572// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1573sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1574 const sp<AudioFlinger::Client>& client, 1575 audio_stream_type_t streamType, 1576 uint32_t sampleRate, 1577 audio_format_t format, 1578 uint32_t channelMask, 1579 int frameCount, 1580 const sp<IMemory>& sharedBuffer, 1581 int sessionId, 1582 bool isTimed, 1583 status_t *status) 1584{ 1585 sp<Track> track; 1586 status_t lStatus; 1587 1588 if (mType == DIRECT) { 1589 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1590 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1591 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1592 "for output %p with format %d", 1593 sampleRate, format, channelMask, mOutput, mFormat); 1594 lStatus = BAD_VALUE; 1595 goto Exit; 1596 } 1597 } 1598 } else { 1599 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1600 if (sampleRate > mSampleRate*2) { 1601 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1602 lStatus = BAD_VALUE; 1603 goto Exit; 1604 } 1605 } 1606 1607 lStatus = initCheck(); 1608 if (lStatus != NO_ERROR) { 1609 ALOGE("Audio driver not initialized."); 1610 goto Exit; 1611 } 1612 1613 { // scope for mLock 1614 Mutex::Autolock _l(mLock); 1615 1616 // all tracks in same audio session must share the same routing strategy otherwise 1617 // conflicts will happen when tracks are moved from one output to another by audio policy 1618 // manager 1619 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1620 for (size_t i = 0; i < mTracks.size(); ++i) { 1621 sp<Track> t = mTracks[i]; 1622 if (t != 0) { 1623 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1624 if (sessionId == t->sessionId() && strategy != actual) { 1625 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1626 strategy, actual); 1627 lStatus = BAD_VALUE; 1628 goto Exit; 1629 } 1630 } 1631 } 1632 1633 if (!isTimed) { 1634 track = new Track(this, client, streamType, sampleRate, format, 1635 channelMask, frameCount, sharedBuffer, sessionId); 1636 } else { 1637 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1638 channelMask, frameCount, sharedBuffer, sessionId); 1639 } 1640 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1641 lStatus = NO_MEMORY; 1642 goto Exit; 1643 } 1644 mTracks.add(track); 1645 1646 sp<EffectChain> chain = getEffectChain_l(sessionId); 1647 if (chain != 0) { 1648 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1649 track->setMainBuffer(chain->inBuffer()); 1650 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1651 chain->incTrackCnt(); 1652 } 1653 1654 // invalidate track immediately if the stream type was moved to another thread since 1655 // createTrack() was called by the client process. 1656 if (!mStreamTypes[streamType].valid) { 1657 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1658 this, streamType); 1659 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1660 } 1661 } 1662 lStatus = NO_ERROR; 1663 1664Exit: 1665 if(status) { 1666 *status = lStatus; 1667 } 1668 return track; 1669} 1670 1671uint32_t AudioFlinger::PlaybackThread::latency() const 1672{ 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() == NO_ERROR) { 1675 return mOutput->stream->get_latency(mOutput->stream); 1676 } else { 1677 return 0; 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 mMasterVolume = value; 1685} 1686 1687void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1688{ 1689 Mutex::Autolock _l(mLock); 1690 setMasterMute_l(muted); 1691} 1692 1693void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1694{ 1695 Mutex::Autolock _l(mLock); 1696 mStreamTypes[stream].volume = value; 1697} 1698 1699void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 mStreamTypes[stream].mute = muted; 1703} 1704 1705float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1706{ 1707 Mutex::Autolock _l(mLock); 1708 return mStreamTypes[stream].volume; 1709} 1710 1711// addTrack_l() must be called with ThreadBase::mLock held 1712status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1713{ 1714 status_t status = ALREADY_EXISTS; 1715 1716 // set retry count for buffer fill 1717 track->mRetryCount = kMaxTrackStartupRetries; 1718 if (mActiveTracks.indexOf(track) < 0) { 1719 // the track is newly added, make sure it fills up all its 1720 // buffers before playing. This is to ensure the client will 1721 // effectively get the latency it requested. 1722 track->mFillingUpStatus = Track::FS_FILLING; 1723 track->mResetDone = false; 1724 mActiveTracks.add(track); 1725 if (track->mainBuffer() != mMixBuffer) { 1726 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1727 if (chain != 0) { 1728 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1729 chain->incActiveTrackCnt(); 1730 } 1731 } 1732 1733 status = NO_ERROR; 1734 } 1735 1736 ALOGV("mWaitWorkCV.broadcast"); 1737 mWaitWorkCV.broadcast(); 1738 1739 return status; 1740} 1741 1742// destroyTrack_l() must be called with ThreadBase::mLock held 1743void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1744{ 1745 track->mState = TrackBase::TERMINATED; 1746 if (mActiveTracks.indexOf(track) < 0) { 1747 removeTrack_l(track); 1748 } 1749} 1750 1751void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1752{ 1753 mTracks.remove(track); 1754 deleteTrackName_l(track->name()); 1755 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1756 if (chain != 0) { 1757 chain->decTrackCnt(); 1758 } 1759} 1760 1761String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1762{ 1763 String8 out_s8 = String8(""); 1764 char *s; 1765 1766 Mutex::Autolock _l(mLock); 1767 if (initCheck() != NO_ERROR) { 1768 return out_s8; 1769 } 1770 1771 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1772 out_s8 = String8(s); 1773 free(s); 1774 return out_s8; 1775} 1776 1777// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1778void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1779 AudioSystem::OutputDescriptor desc; 1780 void *param2 = NULL; 1781 1782 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1783 1784 switch (event) { 1785 case AudioSystem::OUTPUT_OPENED: 1786 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1787 desc.channels = mChannelMask; 1788 desc.samplingRate = mSampleRate; 1789 desc.format = mFormat; 1790 desc.frameCount = mFrameCount; 1791 desc.latency = latency(); 1792 param2 = &desc; 1793 break; 1794 1795 case AudioSystem::STREAM_CONFIG_CHANGED: 1796 param2 = ¶m; 1797 case AudioSystem::OUTPUT_CLOSED: 1798 default: 1799 break; 1800 } 1801 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1802} 1803 1804void AudioFlinger::PlaybackThread::readOutputParameters() 1805{ 1806 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1807 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1808 mChannelCount = (uint16_t)popcount(mChannelMask); 1809 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1811 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1812 1813 // FIXME - Current mixer implementation only supports stereo output: Always 1814 // Allocate a stereo buffer even if HW output is mono. 1815 delete[] mMixBuffer; 1816 mMixBuffer = new int16_t[mFrameCount * 2]; 1817 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1818 1819 // force reconfiguration of effect chains and engines to take new buffer size and audio 1820 // parameters into account 1821 // Note that mLock is not held when readOutputParameters() is called from the constructor 1822 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1823 // matter. 1824 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1825 Vector< sp<EffectChain> > effectChains = mEffectChains; 1826 for (size_t i = 0; i < effectChains.size(); i ++) { 1827 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1828 } 1829} 1830 1831status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1832{ 1833 if (halFrames == NULL || dspFrames == NULL) { 1834 return BAD_VALUE; 1835 } 1836 Mutex::Autolock _l(mLock); 1837 if (initCheck() != NO_ERROR) { 1838 return INVALID_OPERATION; 1839 } 1840 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1841 1842 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1843} 1844 1845uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 uint32_t result = 0; 1849 if (getEffectChain_l(sessionId) != 0) { 1850 result = EFFECT_SESSION; 1851 } 1852 1853 for (size_t i = 0; i < mTracks.size(); ++i) { 1854 sp<Track> track = mTracks[i]; 1855 if (sessionId == track->sessionId() && 1856 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1857 result |= TRACK_SESSION; 1858 break; 1859 } 1860 } 1861 1862 return result; 1863} 1864 1865uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1866{ 1867 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1868 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1869 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1870 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1871 } 1872 for (size_t i = 0; i < mTracks.size(); i++) { 1873 sp<Track> track = mTracks[i]; 1874 if (sessionId == track->sessionId() && 1875 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1876 return AudioSystem::getStrategyForStream(track->streamType()); 1877 } 1878 } 1879 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1880} 1881 1882 1883AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1884{ 1885 Mutex::Autolock _l(mLock); 1886 return mOutput; 1887} 1888 1889AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1890{ 1891 Mutex::Autolock _l(mLock); 1892 AudioStreamOut *output = mOutput; 1893 mOutput = NULL; 1894 return output; 1895} 1896 1897// this method must always be called either with ThreadBase mLock held or inside the thread loop 1898audio_stream_t* AudioFlinger::PlaybackThread::stream() 1899{ 1900 if (mOutput == NULL) { 1901 return NULL; 1902 } 1903 return &mOutput->stream->common; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1907{ 1908 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1909 // decoding and transfer time. So sleeping for half of the latency would likely cause 1910 // underruns 1911 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1912 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1913 } else { 1914 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1915 } 1916} 1917 1918// ---------------------------------------------------------------------------- 1919 1920AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1921 audio_io_handle_t id, uint32_t device, type_t type) 1922 : PlaybackThread(audioFlinger, output, id, device, type) 1923{ 1924 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1925 mPrevMixerStatus = MIXER_IDLE; 1926 // FIXME - Current mixer implementation only supports stereo output 1927 if (mChannelCount == 1) { 1928 ALOGE("Invalid audio hardware channel count"); 1929 } 1930} 1931 1932AudioFlinger::MixerThread::~MixerThread() 1933{ 1934 delete mAudioMixer; 1935} 1936 1937class CpuStats { 1938public: 1939 void sample(); 1940#ifdef DEBUG_CPU_USAGE 1941private: 1942 ThreadCpuUsage mCpu; 1943#endif 1944}; 1945 1946void CpuStats::sample() { 1947#ifdef DEBUG_CPU_USAGE 1948 const CentralTendencyStatistics& stats = mCpu.statistics(); 1949 mCpu.sampleAndEnable(); 1950 unsigned n = stats.n(); 1951 // mCpu.elapsed() is expensive, so don't call it every loop 1952 if ((n & 127) == 1) { 1953 long long elapsed = mCpu.elapsed(); 1954 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1955 double perLoop = elapsed / (double) n; 1956 double perLoop100 = perLoop * 0.01; 1957 double mean = stats.mean(); 1958 double stddev = stats.stddev(); 1959 double minimum = stats.minimum(); 1960 double maximum = stats.maximum(); 1961 mCpu.resetStatistics(); 1962 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1963 elapsed * .000000001, n, perLoop * .000001, 1964 mean * .001, 1965 stddev * .001, 1966 minimum * .001, 1967 maximum * .001, 1968 mean / perLoop100, 1969 stddev / perLoop100, 1970 minimum / perLoop100, 1971 maximum / perLoop100); 1972 } 1973 } 1974#endif 1975}; 1976 1977void AudioFlinger::PlaybackThread::checkSilentMode_l() 1978{ 1979 if (!mMasterMute) { 1980 char value[PROPERTY_VALUE_MAX]; 1981 if (property_get("ro.audio.silent", value, "0") > 0) { 1982 char *endptr; 1983 unsigned long ul = strtoul(value, &endptr, 0); 1984 if (*endptr == '\0' && ul != 0) { 1985 ALOGD("Silence is golden"); 1986 // The setprop command will not allow a property to be changed after 1987 // the first time it is set, so we don't have to worry about un-muting. 1988 setMasterMute_l(true); 1989 } 1990 } 1991 } 1992} 1993 1994bool AudioFlinger::PlaybackThread::threadLoop() 1995{ 1996 // MIXER || DUPLICATING 1997 Vector< sp<Track> > tracksToRemove; 1998 1999 // DIRECT 2000 sp<Track> trackToRemove; 2001 2002 standbyTime = systemTime(); 2003 mixBufferSize = mFrameCount * mFrameSize; 2004 2005 // MIXER 2006 // FIXME: Relaxed timing because of a certain device that can't meet latency 2007 // Should be reduced to 2x after the vendor fixes the driver issue 2008 // increase threshold again due to low power audio mode. The way this warning threshold is 2009 // calculated and its usefulness should be reconsidered anyway. 2010 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2011 nsecs_t lastWarning = 0; 2012if (mType == MIXER) { 2013 longStandbyExit = false; 2014} 2015 2016 // DUPLICATING 2017 // FIXME could this be made local to while loop? 2018 writeFrames = 0; 2019 2020 activeSleepTime = activeSleepTimeUs(); 2021 idleSleepTime = idleSleepTimeUs(); 2022 sleepTime = idleSleepTime; 2023 2024if (mType == MIXER) { 2025 sleepTimeShift = 0; 2026} 2027 2028 // MIXER 2029 CpuStats cpuStats; 2030 2031 // DIRECT 2032if (mType == DIRECT) { 2033 // use shorter standby delay as on normal output to release 2034 // hardware resources as soon as possible 2035 standbyDelay = microseconds(activeSleepTime*2); 2036} 2037 2038 acquireWakeLock(); 2039 2040 while (!exitPending()) 2041 { 2042if (mType == MIXER) { 2043 cpuStats.sample(); 2044} 2045 2046 Vector< sp<EffectChain> > effectChains; 2047 2048 processConfigEvents(); 2049 2050if (mType == DIRECT) { 2051 activeTrack.clear(); 2052} 2053 2054 mixerStatus = MIXER_IDLE; 2055 { // scope for mLock 2056 2057 Mutex::Autolock _l(mLock); 2058 2059 if (checkForNewParameters_l()) { 2060 mixBufferSize = mFrameCount * mFrameSize; 2061 2062if (mType == MIXER) { 2063 // FIXME: Relaxed timing because of a certain device that can't meet latency 2064 // Should be reduced to 2x after the vendor fixes the driver issue 2065 // increase threshold again due to low power audio mode. The way this warning 2066 // threshold is calculated and its usefulness should be reconsidered anyway. 2067 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2068} 2069 2070if (mType == DUPLICATING) { 2071 updateWaitTime(); 2072} 2073 2074 activeSleepTime = activeSleepTimeUs(); 2075 idleSleepTime = idleSleepTimeUs(); 2076 2077if (mType == DIRECT) { 2078 standbyDelay = microseconds(activeSleepTime*2); 2079} 2080 2081 } 2082 2083if (mType == DUPLICATING) { 2084#if 0 // see earlier FIXME 2085 // Now that this is a field instead of local variable, 2086 // clear it so it is empty the first time through the loop, 2087 // and later an assignment could combine the clear with the loop below 2088 outputTracks.clear(); 2089#endif 2090 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2091 outputTracks.add(mOutputTracks[i]); 2092 } 2093} 2094 2095 // put audio hardware into standby after short delay 2096 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2097 mSuspended > 0)) { 2098 if (!mStandby) { 2099 2100 threadLoop_standby(); 2101 2102 mStandby = true; 2103 mBytesWritten = 0; 2104 } 2105 2106 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2107 // we're about to wait, flush the binder command buffer 2108 IPCThreadState::self()->flushCommands(); 2109 2110if (mType == DUPLICATING) { 2111 outputTracks.clear(); 2112} 2113 2114 if (exitPending()) break; 2115 2116 releaseWakeLock_l(); 2117 // wait until we have something to do... 2118 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2119 mWaitWorkCV.wait(mLock); 2120 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2121 acquireWakeLock_l(); 2122 2123if (mType == MIXER || mType == DUPLICATING) { 2124 mPrevMixerStatus = MIXER_IDLE; 2125} 2126 2127 checkSilentMode_l(); 2128 2129if (mType == MIXER || mType == DUPLICATING) { 2130 standbyTime = systemTime() + mStandbyTimeInNsecs; 2131} 2132 2133if (mType == DIRECT) { 2134 standbyTime = systemTime() + standbyDelay; 2135} 2136 2137 sleepTime = idleSleepTime; 2138 2139if (mType == MIXER) { 2140 sleepTimeShift = 0; 2141} 2142 2143 continue; 2144 } 2145 } 2146 2147// FIXME merge these 2148if (mType == MIXER || mType == DUPLICATING) { 2149 mixerStatus = prepareTracks_l(&tracksToRemove); 2150} 2151if (mType == DIRECT) { 2152 mixerStatus = threadLoop_prepareTracks_l(trackToRemove); 2153 // see FIXME in AudioFlinger.h 2154 if (mixerStatus == MIXER_CONTINUE) { 2155 continue; 2156 } 2157} 2158 2159 // prevent any changes in effect chain list and in each effect chain 2160 // during mixing and effect process as the audio buffers could be deleted 2161 // or modified if an effect is created or deleted 2162 lockEffectChains_l(effectChains); 2163 } 2164 2165if (mType == DIRECT) { 2166 // For DirectOutputThread, this test is equivalent to "activeTrack != 0" 2167} 2168 2169 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2170 threadLoop_mix(); 2171 } else { 2172 threadLoop_sleepTime(); 2173 } 2174 2175 if (mSuspended > 0) { 2176 sleepTime = suspendSleepTimeUs(); 2177 } 2178 2179 // only process effects if we're going to write 2180 if (sleepTime == 0) { 2181 2182 if (mixerStatus == MIXER_TRACKS_READY) { 2183 2184 // Non-trivial for DIRECT only 2185 applyVolume(); 2186 2187 } 2188 2189 for (size_t i = 0; i < effectChains.size(); i ++) { 2190 effectChains[i]->process_l(); 2191 } 2192 } 2193 2194 // enable changes in effect chain 2195 unlockEffectChains(effectChains); 2196 2197 // sleepTime == 0 means we must write to audio hardware 2198 if (sleepTime == 0) { 2199 2200 threadLoop_write(); 2201 2202if (mType == MIXER) { 2203 // write blocked detection 2204 nsecs_t now = systemTime(); 2205 nsecs_t delta = now - mLastWriteTime; 2206 if (!mStandby && delta > maxPeriod) { 2207 mNumDelayedWrites++; 2208 if ((now - lastWarning) > kWarningThrottleNs) { 2209 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2210 ns2ms(delta), mNumDelayedWrites, this); 2211 lastWarning = now; 2212 } 2213 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2214 // a different threshold. Or completely removed for what it is worth anyway... 2215 if (mStandby) { 2216 longStandbyExit = true; 2217 } 2218 } 2219} 2220 2221 mStandby = false; 2222 } else { 2223 usleep(sleepTime); 2224 } 2225 2226 // finally let go of removed track(s), without the lock held 2227 // since we can't guarantee the destructors won't acquire that 2228 // same lock. 2229 2230// FIXME merge these 2231if (mType == MIXER) { 2232 tracksToRemove.clear(); 2233} 2234if (mType == DIRECT) { 2235 trackToRemove.clear(); 2236 activeTrack.clear(); 2237} 2238if (mType == DUPLICATING) { 2239 tracksToRemove.clear(); 2240 outputTracks.clear(); 2241} 2242 2243 // Effect chains will be actually deleted here if they were removed from 2244 // mEffectChains list during mixing or effects processing 2245 effectChains.clear(); 2246 2247 // FIXME Note that the above .clear() is no longer necessary since effectChains 2248 // is now local to this block, but will keep it for now (at least until merge done). 2249 } 2250 2251if (mType == MIXER || mType == DIRECT) { 2252 // put output stream into standby mode 2253 if (!mStandby) { 2254 mOutput->stream->common.standby(&mOutput->stream->common); 2255 } 2256} 2257if (mType == DUPLICATING) { 2258 // for DuplicatingThread, standby mode is handled by the outputTracks 2259} 2260 2261 releaseWakeLock(); 2262 2263 ALOGV("Thread %p type %d exiting", this, mType); 2264 return false; 2265} 2266 2267// shared by MIXER and DIRECT, overridden by DUPLICATING 2268void AudioFlinger::PlaybackThread::threadLoop_write() 2269{ 2270 // FIXME rewrite to reduce number of system calls 2271 mLastWriteTime = systemTime(); 2272 mInWrite = true; 2273 mBytesWritten += mixBufferSize; 2274 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2275 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2276 mNumWrites++; 2277 mInWrite = false; 2278} 2279 2280// shared by MIXER and DIRECT, overridden by DUPLICATING 2281void AudioFlinger::PlaybackThread::threadLoop_standby() 2282{ 2283 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2284 mOutput->stream->common.standby(&mOutput->stream->common); 2285} 2286 2287void AudioFlinger::MixerThread::threadLoop_mix() 2288{ 2289 // obtain the presentation timestamp of the next output buffer 2290 int64_t pts; 2291 status_t status = INVALID_OPERATION; 2292 2293 if (NULL != mOutput->stream->get_next_write_timestamp) { 2294 status = mOutput->stream->get_next_write_timestamp( 2295 mOutput->stream, &pts); 2296 } 2297 2298 if (status != NO_ERROR) { 2299 pts = AudioBufferProvider::kInvalidPTS; 2300 } 2301 2302 // mix buffers... 2303 mAudioMixer->process(pts); 2304 // increase sleep time progressively when application underrun condition clears. 2305 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2306 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2307 // such that we would underrun the audio HAL. 2308 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2309 sleepTimeShift--; 2310 } 2311 sleepTime = 0; 2312 standbyTime = systemTime() + mStandbyTimeInNsecs; 2313 //TODO: delay standby when effects have a tail 2314} 2315 2316void AudioFlinger::MixerThread::threadLoop_sleepTime() 2317{ 2318 // If no tracks are ready, sleep once for the duration of an output 2319 // buffer size, then write 0s to the output 2320 if (sleepTime == 0) { 2321 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2322 sleepTime = activeSleepTime >> sleepTimeShift; 2323 if (sleepTime < kMinThreadSleepTimeUs) { 2324 sleepTime = kMinThreadSleepTimeUs; 2325 } 2326 // reduce sleep time in case of consecutive application underruns to avoid 2327 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2328 // duration we would end up writing less data than needed by the audio HAL if 2329 // the condition persists. 2330 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2331 sleepTimeShift++; 2332 } 2333 } else { 2334 sleepTime = idleSleepTime; 2335 } 2336 } else if (mBytesWritten != 0 || 2337 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2338 memset (mMixBuffer, 0, mixBufferSize); 2339 sleepTime = 0; 2340 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2341 } 2342 // TODO add standby time extension fct of effect tail 2343} 2344 2345// prepareTracks_l() must be called with ThreadBase::mLock held 2346AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2347 Vector< sp<Track> > *tracksToRemove) 2348{ 2349 2350 mixer_state mixerStatus = MIXER_IDLE; 2351 // find out which tracks need to be processed 2352 size_t count = mActiveTracks.size(); 2353 size_t mixedTracks = 0; 2354 size_t tracksWithEffect = 0; 2355 2356 float masterVolume = mMasterVolume; 2357 bool masterMute = mMasterMute; 2358 2359 if (masterMute) { 2360 masterVolume = 0; 2361 } 2362 // Delegate master volume control to effect in output mix effect chain if needed 2363 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2364 if (chain != 0) { 2365 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2366 chain->setVolume_l(&v, &v); 2367 masterVolume = (float)((v + (1 << 23)) >> 24); 2368 chain.clear(); 2369 } 2370 2371 for (size_t i=0 ; i<count ; i++) { 2372 sp<Track> t = mActiveTracks[i].promote(); 2373 if (t == 0) continue; 2374 2375 // this const just means the local variable doesn't change 2376 Track* const track = t.get(); 2377 audio_track_cblk_t* cblk = track->cblk(); 2378 2379 // The first time a track is added we wait 2380 // for all its buffers to be filled before processing it 2381 int name = track->name(); 2382 // make sure that we have enough frames to mix one full buffer. 2383 // enforce this condition only once to enable draining the buffer in case the client 2384 // app does not call stop() and relies on underrun to stop: 2385 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2386 // during last round 2387 uint32_t minFrames = 1; 2388 if (!track->isStopped() && !track->isPausing() && 2389 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2390 if (t->sampleRate() == (int)mSampleRate) { 2391 minFrames = mFrameCount; 2392 } else { 2393 // +1 for rounding and +1 for additional sample needed for interpolation 2394 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2395 // add frames already consumed but not yet released by the resampler 2396 // because cblk->framesReady() will include these frames 2397 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2398 // the minimum track buffer size is normally twice the number of frames necessary 2399 // to fill one buffer and the resampler should not leave more than one buffer worth 2400 // of unreleased frames after each pass, but just in case... 2401 ALOG_ASSERT(minFrames <= cblk->frameCount); 2402 } 2403 } 2404 if ((track->framesReady() >= minFrames) && track->isReady() && 2405 !track->isPaused() && !track->isTerminated()) 2406 { 2407 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2408 2409 mixedTracks++; 2410 2411 // track->mainBuffer() != mMixBuffer means there is an effect chain 2412 // connected to the track 2413 chain.clear(); 2414 if (track->mainBuffer() != mMixBuffer) { 2415 chain = getEffectChain_l(track->sessionId()); 2416 // Delegate volume control to effect in track effect chain if needed 2417 if (chain != 0) { 2418 tracksWithEffect++; 2419 } else { 2420 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2421 name, track->sessionId()); 2422 } 2423 } 2424 2425 2426 int param = AudioMixer::VOLUME; 2427 if (track->mFillingUpStatus == Track::FS_FILLED) { 2428 // no ramp for the first volume setting 2429 track->mFillingUpStatus = Track::FS_ACTIVE; 2430 if (track->mState == TrackBase::RESUMING) { 2431 track->mState = TrackBase::ACTIVE; 2432 param = AudioMixer::RAMP_VOLUME; 2433 } 2434 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2435 } else if (cblk->server != 0) { 2436 // If the track is stopped before the first frame was mixed, 2437 // do not apply ramp 2438 param = AudioMixer::RAMP_VOLUME; 2439 } 2440 2441 // compute volume for this track 2442 uint32_t vl, vr, va; 2443 if (track->isMuted() || track->isPausing() || 2444 mStreamTypes[track->streamType()].mute) { 2445 vl = vr = va = 0; 2446 if (track->isPausing()) { 2447 track->setPaused(); 2448 } 2449 } else { 2450 2451 // read original volumes with volume control 2452 float typeVolume = mStreamTypes[track->streamType()].volume; 2453 float v = masterVolume * typeVolume; 2454 uint32_t vlr = cblk->getVolumeLR(); 2455 vl = vlr & 0xFFFF; 2456 vr = vlr >> 16; 2457 // track volumes come from shared memory, so can't be trusted and must be clamped 2458 if (vl > MAX_GAIN_INT) { 2459 ALOGV("Track left volume out of range: %04X", vl); 2460 vl = MAX_GAIN_INT; 2461 } 2462 if (vr > MAX_GAIN_INT) { 2463 ALOGV("Track right volume out of range: %04X", vr); 2464 vr = MAX_GAIN_INT; 2465 } 2466 // now apply the master volume and stream type volume 2467 vl = (uint32_t)(v * vl) << 12; 2468 vr = (uint32_t)(v * vr) << 12; 2469 // assuming master volume and stream type volume each go up to 1.0, 2470 // vl and vr are now in 8.24 format 2471 2472 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2473 // send level comes from shared memory and so may be corrupt 2474 if (sendLevel > MAX_GAIN_INT) { 2475 ALOGV("Track send level out of range: %04X", sendLevel); 2476 sendLevel = MAX_GAIN_INT; 2477 } 2478 va = (uint32_t)(v * sendLevel); 2479 } 2480 // Delegate volume control to effect in track effect chain if needed 2481 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2482 // Do not ramp volume if volume is controlled by effect 2483 param = AudioMixer::VOLUME; 2484 track->mHasVolumeController = true; 2485 } else { 2486 // force no volume ramp when volume controller was just disabled or removed 2487 // from effect chain to avoid volume spike 2488 if (track->mHasVolumeController) { 2489 param = AudioMixer::VOLUME; 2490 } 2491 track->mHasVolumeController = false; 2492 } 2493 2494 // Convert volumes from 8.24 to 4.12 format 2495 // This additional clamping is needed in case chain->setVolume_l() overshot 2496 vl = (vl + (1 << 11)) >> 12; 2497 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2498 vr = (vr + (1 << 11)) >> 12; 2499 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2500 2501 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2502 2503 // XXX: these things DON'T need to be done each time 2504 mAudioMixer->setBufferProvider(name, track); 2505 mAudioMixer->enable(name); 2506 2507 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2508 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2509 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2510 mAudioMixer->setParameter( 2511 name, 2512 AudioMixer::TRACK, 2513 AudioMixer::FORMAT, (void *)track->format()); 2514 mAudioMixer->setParameter( 2515 name, 2516 AudioMixer::TRACK, 2517 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2518 mAudioMixer->setParameter( 2519 name, 2520 AudioMixer::RESAMPLE, 2521 AudioMixer::SAMPLE_RATE, 2522 (void *)(cblk->sampleRate)); 2523 mAudioMixer->setParameter( 2524 name, 2525 AudioMixer::TRACK, 2526 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2527 mAudioMixer->setParameter( 2528 name, 2529 AudioMixer::TRACK, 2530 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2531 2532 // reset retry count 2533 track->mRetryCount = kMaxTrackRetries; 2534 // If one track is ready, set the mixer ready if: 2535 // - the mixer was not ready during previous round OR 2536 // - no other track is not ready 2537 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2538 mixerStatus != MIXER_TRACKS_ENABLED) { 2539 mixerStatus = MIXER_TRACKS_READY; 2540 } 2541 } else { 2542 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2543 if (track->isStopped()) { 2544 track->reset(); 2545 } 2546 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2547 // We have consumed all the buffers of this track. 2548 // Remove it from the list of active tracks. 2549 tracksToRemove->add(track); 2550 } else { 2551 // No buffers for this track. Give it a few chances to 2552 // fill a buffer, then remove it from active list. 2553 if (--(track->mRetryCount) <= 0) { 2554 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2555 tracksToRemove->add(track); 2556 // indicate to client process that the track was disabled because of underrun 2557 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2558 // If one track is not ready, mark the mixer also not ready if: 2559 // - the mixer was ready during previous round OR 2560 // - no other track is ready 2561 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2562 mixerStatus != MIXER_TRACKS_READY) { 2563 mixerStatus = MIXER_TRACKS_ENABLED; 2564 } 2565 } 2566 mAudioMixer->disable(name); 2567 } 2568 } 2569 2570 // remove all the tracks that need to be... 2571 count = tracksToRemove->size(); 2572 if (CC_UNLIKELY(count)) { 2573 for (size_t i=0 ; i<count ; i++) { 2574 const sp<Track>& track = tracksToRemove->itemAt(i); 2575 mActiveTracks.remove(track); 2576 if (track->mainBuffer() != mMixBuffer) { 2577 chain = getEffectChain_l(track->sessionId()); 2578 if (chain != 0) { 2579 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2580 chain->decActiveTrackCnt(); 2581 } 2582 } 2583 if (track->isTerminated()) { 2584 removeTrack_l(track); 2585 } 2586 } 2587 } 2588 2589 // mix buffer must be cleared if all tracks are connected to an 2590 // effect chain as in this case the mixer will not write to 2591 // mix buffer and track effects will accumulate into it 2592 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2593 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2594 } 2595 2596 mPrevMixerStatus = mixerStatus; 2597 return mixerStatus; 2598} 2599 2600void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2601{ 2602 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2603 this, streamType, mTracks.size()); 2604 Mutex::Autolock _l(mLock); 2605 2606 size_t size = mTracks.size(); 2607 for (size_t i = 0; i < size; i++) { 2608 sp<Track> t = mTracks[i]; 2609 if (t->streamType() == streamType) { 2610 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2611 t->mCblk->cv.signal(); 2612 } 2613 } 2614} 2615 2616void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2617{ 2618 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2619 this, streamType, valid); 2620 Mutex::Autolock _l(mLock); 2621 2622 mStreamTypes[streamType].valid = valid; 2623} 2624 2625// getTrackName_l() must be called with ThreadBase::mLock held 2626int AudioFlinger::MixerThread::getTrackName_l() 2627{ 2628 return mAudioMixer->getTrackName(); 2629} 2630 2631// deleteTrackName_l() must be called with ThreadBase::mLock held 2632void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2633{ 2634 ALOGV("remove track (%d) and delete from mixer", name); 2635 mAudioMixer->deleteTrackName(name); 2636} 2637 2638// checkForNewParameters_l() must be called with ThreadBase::mLock held 2639bool AudioFlinger::MixerThread::checkForNewParameters_l() 2640{ 2641 bool reconfig = false; 2642 2643 while (!mNewParameters.isEmpty()) { 2644 status_t status = NO_ERROR; 2645 String8 keyValuePair = mNewParameters[0]; 2646 AudioParameter param = AudioParameter(keyValuePair); 2647 int value; 2648 2649 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2650 reconfig = true; 2651 } 2652 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2653 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2654 status = BAD_VALUE; 2655 } else { 2656 reconfig = true; 2657 } 2658 } 2659 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2660 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2661 status = BAD_VALUE; 2662 } else { 2663 reconfig = true; 2664 } 2665 } 2666 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2667 // do not accept frame count changes if tracks are open as the track buffer 2668 // size depends on frame count and correct behavior would not be guaranteed 2669 // if frame count is changed after track creation 2670 if (!mTracks.isEmpty()) { 2671 status = INVALID_OPERATION; 2672 } else { 2673 reconfig = true; 2674 } 2675 } 2676 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2677 // when changing the audio output device, call addBatteryData to notify 2678 // the change 2679 if ((int)mDevice != value) { 2680 uint32_t params = 0; 2681 // check whether speaker is on 2682 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2683 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2684 } 2685 2686 int deviceWithoutSpeaker 2687 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2688 // check if any other device (except speaker) is on 2689 if (value & deviceWithoutSpeaker ) { 2690 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2691 } 2692 2693 if (params != 0) { 2694 addBatteryData(params); 2695 } 2696 } 2697 2698 // forward device change to effects that have requested to be 2699 // aware of attached audio device. 2700 mDevice = (uint32_t)value; 2701 for (size_t i = 0; i < mEffectChains.size(); i++) { 2702 mEffectChains[i]->setDevice_l(mDevice); 2703 } 2704 } 2705 2706 if (status == NO_ERROR) { 2707 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2708 keyValuePair.string()); 2709 if (!mStandby && status == INVALID_OPERATION) { 2710 mOutput->stream->common.standby(&mOutput->stream->common); 2711 mStandby = true; 2712 mBytesWritten = 0; 2713 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2714 keyValuePair.string()); 2715 } 2716 if (status == NO_ERROR && reconfig) { 2717 delete mAudioMixer; 2718 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2719 mAudioMixer = NULL; 2720 readOutputParameters(); 2721 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2722 for (size_t i = 0; i < mTracks.size() ; i++) { 2723 int name = getTrackName_l(); 2724 if (name < 0) break; 2725 mTracks[i]->mName = name; 2726 // limit track sample rate to 2 x new output sample rate 2727 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2728 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2729 } 2730 } 2731 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2732 } 2733 } 2734 2735 mNewParameters.removeAt(0); 2736 2737 mParamStatus = status; 2738 mParamCond.signal(); 2739 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2740 // already timed out waiting for the status and will never signal the condition. 2741 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2742 } 2743 return reconfig; 2744} 2745 2746status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2747{ 2748 const size_t SIZE = 256; 2749 char buffer[SIZE]; 2750 String8 result; 2751 2752 PlaybackThread::dumpInternals(fd, args); 2753 2754 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2755 result.append(buffer); 2756 write(fd, result.string(), result.size()); 2757 return NO_ERROR; 2758} 2759 2760uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2761{ 2762 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2763} 2764 2765uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2766{ 2767 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2768} 2769 2770// ---------------------------------------------------------------------------- 2771AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2772 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2773 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2774 // mLeftVolFloat, mRightVolFloat 2775 // mLeftVolShort, mRightVolShort 2776{ 2777} 2778 2779AudioFlinger::DirectOutputThread::~DirectOutputThread() 2780{ 2781} 2782 2783void AudioFlinger::DirectOutputThread::applyVolume() 2784{ 2785 // Do not apply volume on compressed audio 2786 if (!audio_is_linear_pcm(mFormat)) { 2787 return; 2788 } 2789 2790 // convert to signed 16 bit before volume calculation 2791 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2792 size_t count = mFrameCount * mChannelCount; 2793 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2794 int16_t *dst = mMixBuffer + count-1; 2795 while(count--) { 2796 *dst-- = (int16_t)(*src--^0x80) << 8; 2797 } 2798 } 2799 2800 size_t frameCount = mFrameCount; 2801 int16_t *out = mMixBuffer; 2802 if (rampVolume) { 2803 if (mChannelCount == 1) { 2804 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2805 int32_t vlInc = d / (int32_t)frameCount; 2806 int32_t vl = ((int32_t)mLeftVolShort << 16); 2807 do { 2808 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2809 out++; 2810 vl += vlInc; 2811 } while (--frameCount); 2812 2813 } else { 2814 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2815 int32_t vlInc = d / (int32_t)frameCount; 2816 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2817 int32_t vrInc = d / (int32_t)frameCount; 2818 int32_t vl = ((int32_t)mLeftVolShort << 16); 2819 int32_t vr = ((int32_t)mRightVolShort << 16); 2820 do { 2821 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2822 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2823 out += 2; 2824 vl += vlInc; 2825 vr += vrInc; 2826 } while (--frameCount); 2827 } 2828 } else { 2829 if (mChannelCount == 1) { 2830 do { 2831 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2832 out++; 2833 } while (--frameCount); 2834 } else { 2835 do { 2836 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2837 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2838 out += 2; 2839 } while (--frameCount); 2840 } 2841 } 2842 2843 // convert back to unsigned 8 bit after volume calculation 2844 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2845 size_t count = mFrameCount * mChannelCount; 2846 int16_t *src = mMixBuffer; 2847 uint8_t *dst = (uint8_t *)mMixBuffer; 2848 while(count--) { 2849 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2850 } 2851 } 2852 2853 mLeftVolShort = leftVol; 2854 mRightVolShort = rightVol; 2855} 2856 2857AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::threadLoop_prepareTracks_l( 2858 sp<Track>& trackToRemove 2859) 2860{ 2861// FIXME Temporarily renamed to avoid confusion with the member "mixerStatus" 2862mixer_state mixerStatus_ = MIXER_IDLE; 2863 2864 // find out which tracks need to be processed 2865 if (mActiveTracks.size() != 0) { 2866 sp<Track> t = mActiveTracks[0].promote(); 2867 // see FIXME in AudioFlinger.h, return MIXER_IDLE might also work 2868 if (t == 0) return MIXER_CONTINUE; 2869 //if (t == 0) continue; 2870 2871 Track* const track = t.get(); 2872 audio_track_cblk_t* cblk = track->cblk(); 2873 2874 // The first time a track is added we wait 2875 // for all its buffers to be filled before processing it 2876 if (cblk->framesReady() && track->isReady() && 2877 !track->isPaused() && !track->isTerminated()) 2878 { 2879 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2880 2881 if (track->mFillingUpStatus == Track::FS_FILLED) { 2882 track->mFillingUpStatus = Track::FS_ACTIVE; 2883 mLeftVolFloat = mRightVolFloat = 0; 2884 mLeftVolShort = mRightVolShort = 0; 2885 if (track->mState == TrackBase::RESUMING) { 2886 track->mState = TrackBase::ACTIVE; 2887 rampVolume = true; 2888 } 2889 } else if (cblk->server != 0) { 2890 // If the track is stopped before the first frame was mixed, 2891 // do not apply ramp 2892 rampVolume = true; 2893 } 2894 // compute volume for this track 2895 float left, right; 2896 if (track->isMuted() || mMasterMute || track->isPausing() || 2897 mStreamTypes[track->streamType()].mute) { 2898 left = right = 0; 2899 if (track->isPausing()) { 2900 track->setPaused(); 2901 } 2902 } else { 2903 float typeVolume = mStreamTypes[track->streamType()].volume; 2904 float v = mMasterVolume * typeVolume; 2905 uint32_t vlr = cblk->getVolumeLR(); 2906 float v_clamped = v * (vlr & 0xFFFF); 2907 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2908 left = v_clamped/MAX_GAIN; 2909 v_clamped = v * (vlr >> 16); 2910 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2911 right = v_clamped/MAX_GAIN; 2912 } 2913 2914 if (left != mLeftVolFloat || right != mRightVolFloat) { 2915 mLeftVolFloat = left; 2916 mRightVolFloat = right; 2917 2918 // If audio HAL implements volume control, 2919 // force software volume to nominal value 2920 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2921 left = 1.0f; 2922 right = 1.0f; 2923 } 2924 2925 // Convert volumes from float to 8.24 2926 uint32_t vl = (uint32_t)(left * (1 << 24)); 2927 uint32_t vr = (uint32_t)(right * (1 << 24)); 2928 2929 // Delegate volume control to effect in track effect chain if needed 2930 // only one effect chain can be present on DirectOutputThread, so if 2931 // there is one, the track is connected to it 2932 if (!mEffectChains.isEmpty()) { 2933 // Do not ramp volume if volume is controlled by effect 2934 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2935 rampVolume = false; 2936 } 2937 } 2938 2939 // Convert volumes from 8.24 to 4.12 format 2940 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2941 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2942 leftVol = (uint16_t)v_clamped; 2943 v_clamped = (vr + (1 << 11)) >> 12; 2944 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2945 rightVol = (uint16_t)v_clamped; 2946 } else { 2947 leftVol = mLeftVolShort; 2948 rightVol = mRightVolShort; 2949 rampVolume = false; 2950 } 2951 2952 // reset retry count 2953 track->mRetryCount = kMaxTrackRetriesDirect; 2954 activeTrack = t; 2955 mixerStatus_ = MIXER_TRACKS_READY; 2956 } else { 2957 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2958 if (track->isStopped()) { 2959 track->reset(); 2960 } 2961 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2962 // We have consumed all the buffers of this track. 2963 // Remove it from the list of active tracks. 2964 trackToRemove = track; 2965 } else { 2966 // No buffers for this track. Give it a few chances to 2967 // fill a buffer, then remove it from active list. 2968 if (--(track->mRetryCount) <= 0) { 2969 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2970 trackToRemove = track; 2971 } else { 2972 mixerStatus_ = MIXER_TRACKS_ENABLED; 2973 } 2974 } 2975 } 2976 } 2977 2978 // remove all the tracks that need to be... 2979 if (CC_UNLIKELY(trackToRemove != 0)) { 2980 mActiveTracks.remove(trackToRemove); 2981 if (!mEffectChains.isEmpty()) { 2982 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2983 trackToRemove->sessionId()); 2984 mEffectChains[0]->decActiveTrackCnt(); 2985 } 2986 if (trackToRemove->isTerminated()) { 2987 removeTrack_l(trackToRemove); 2988 } 2989 } 2990 2991return mixerStatus_; 2992} 2993 2994void AudioFlinger::DirectOutputThread::threadLoop_mix() 2995{ 2996 AudioBufferProvider::Buffer buffer; 2997 size_t frameCount = mFrameCount; 2998 int8_t *curBuf = (int8_t *)mMixBuffer; 2999 // output audio to hardware 3000 while (frameCount) { 3001 buffer.frameCount = frameCount; 3002 activeTrack->getNextBuffer(&buffer); 3003 if (CC_UNLIKELY(buffer.raw == NULL)) { 3004 memset(curBuf, 0, frameCount * mFrameSize); 3005 break; 3006 } 3007 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3008 frameCount -= buffer.frameCount; 3009 curBuf += buffer.frameCount * mFrameSize; 3010 activeTrack->releaseBuffer(&buffer); 3011 } 3012 sleepTime = 0; 3013 standbyTime = systemTime() + standbyDelay; 3014} 3015 3016void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3017{ 3018 if (sleepTime == 0) { 3019 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3020 sleepTime = activeSleepTime; 3021 } else { 3022 sleepTime = idleSleepTime; 3023 } 3024 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3025 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3026 sleepTime = 0; 3027 } 3028} 3029 3030// getTrackName_l() must be called with ThreadBase::mLock held 3031int AudioFlinger::DirectOutputThread::getTrackName_l() 3032{ 3033 return 0; 3034} 3035 3036// deleteTrackName_l() must be called with ThreadBase::mLock held 3037void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3038{ 3039} 3040 3041// checkForNewParameters_l() must be called with ThreadBase::mLock held 3042bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3043{ 3044 bool reconfig = false; 3045 3046 while (!mNewParameters.isEmpty()) { 3047 status_t status = NO_ERROR; 3048 String8 keyValuePair = mNewParameters[0]; 3049 AudioParameter param = AudioParameter(keyValuePair); 3050 int value; 3051 3052 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3053 // do not accept frame count changes if tracks are open as the track buffer 3054 // size depends on frame count and correct behavior would not be garantied 3055 // if frame count is changed after track creation 3056 if (!mTracks.isEmpty()) { 3057 status = INVALID_OPERATION; 3058 } else { 3059 reconfig = true; 3060 } 3061 } 3062 if (status == NO_ERROR) { 3063 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3064 keyValuePair.string()); 3065 if (!mStandby && status == INVALID_OPERATION) { 3066 mOutput->stream->common.standby(&mOutput->stream->common); 3067 mStandby = true; 3068 mBytesWritten = 0; 3069 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3070 keyValuePair.string()); 3071 } 3072 if (status == NO_ERROR && reconfig) { 3073 readOutputParameters(); 3074 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3075 } 3076 } 3077 3078 mNewParameters.removeAt(0); 3079 3080 mParamStatus = status; 3081 mParamCond.signal(); 3082 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3083 // already timed out waiting for the status and will never signal the condition. 3084 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3085 } 3086 return reconfig; 3087} 3088 3089uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3090{ 3091 uint32_t time; 3092 if (audio_is_linear_pcm(mFormat)) { 3093 time = PlaybackThread::activeSleepTimeUs(); 3094 } else { 3095 time = 10000; 3096 } 3097 return time; 3098} 3099 3100uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3101{ 3102 uint32_t time; 3103 if (audio_is_linear_pcm(mFormat)) { 3104 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3105 } else { 3106 time = 10000; 3107 } 3108 return time; 3109} 3110 3111uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3112{ 3113 uint32_t time; 3114 if (audio_is_linear_pcm(mFormat)) { 3115 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3116 } else { 3117 time = 10000; 3118 } 3119 return time; 3120} 3121 3122 3123// ---------------------------------------------------------------------------- 3124 3125AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3126 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3127 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3128 mWaitTimeMs(UINT_MAX) 3129{ 3130 addOutputTrack(mainThread); 3131} 3132 3133AudioFlinger::DuplicatingThread::~DuplicatingThread() 3134{ 3135 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3136 mOutputTracks[i]->destroy(); 3137 } 3138} 3139 3140void AudioFlinger::DuplicatingThread::threadLoop_mix() 3141{ 3142 // mix buffers... 3143 if (outputsReady(outputTracks)) { 3144 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3145 } else { 3146 memset(mMixBuffer, 0, mixBufferSize); 3147 } 3148 sleepTime = 0; 3149 writeFrames = mFrameCount; 3150} 3151 3152void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3153{ 3154 if (sleepTime == 0) { 3155 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3156 sleepTime = activeSleepTime; 3157 } else { 3158 sleepTime = idleSleepTime; 3159 } 3160 } else if (mBytesWritten != 0) { 3161 // flush remaining overflow buffers in output tracks 3162 for (size_t i = 0; i < outputTracks.size(); i++) { 3163 if (outputTracks[i]->isActive()) { 3164 sleepTime = 0; 3165 writeFrames = 0; 3166 memset(mMixBuffer, 0, mixBufferSize); 3167 break; 3168 } 3169 } 3170 } 3171} 3172 3173void AudioFlinger::DuplicatingThread::threadLoop_write() 3174{ 3175 standbyTime = systemTime() + mStandbyTimeInNsecs; 3176 for (size_t i = 0; i < outputTracks.size(); i++) { 3177 outputTracks[i]->write(mMixBuffer, writeFrames); 3178 } 3179 mBytesWritten += mixBufferSize; 3180} 3181 3182void AudioFlinger::DuplicatingThread::threadLoop_standby() 3183{ 3184 // DuplicatingThread implements standby by stopping all tracks 3185 for (size_t i = 0; i < outputTracks.size(); i++) { 3186 outputTracks[i]->stop(); 3187 } 3188} 3189 3190void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3191{ 3192 Mutex::Autolock _l(mLock); 3193 // FIXME explain this formula 3194 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3195 OutputTrack *outputTrack = new OutputTrack(thread, 3196 this, 3197 mSampleRate, 3198 mFormat, 3199 mChannelMask, 3200 frameCount); 3201 if (outputTrack->cblk() != NULL) { 3202 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3203 mOutputTracks.add(outputTrack); 3204 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3205 updateWaitTime(); 3206 } 3207} 3208 3209void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3210{ 3211 Mutex::Autolock _l(mLock); 3212 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3213 if (mOutputTracks[i]->thread() == thread) { 3214 mOutputTracks[i]->destroy(); 3215 mOutputTracks.removeAt(i); 3216 updateWaitTime(); 3217 return; 3218 } 3219 } 3220 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3221} 3222 3223void AudioFlinger::DuplicatingThread::updateWaitTime() 3224{ 3225 mWaitTimeMs = UINT_MAX; 3226 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3227 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3228 if (strong != 0) { 3229 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3230 if (waitTimeMs < mWaitTimeMs) { 3231 mWaitTimeMs = waitTimeMs; 3232 } 3233 } 3234 } 3235} 3236 3237 3238bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3239{ 3240 for (size_t i = 0; i < outputTracks.size(); i++) { 3241 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3242 if (thread == 0) { 3243 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3244 return false; 3245 } 3246 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3247 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3248 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3249 return false; 3250 } 3251 } 3252 return true; 3253} 3254 3255uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3256{ 3257 return (mWaitTimeMs * 1000) / 2; 3258} 3259 3260// ---------------------------------------------------------------------------- 3261 3262// TrackBase constructor must be called with AudioFlinger::mLock held 3263AudioFlinger::ThreadBase::TrackBase::TrackBase( 3264 ThreadBase *thread, 3265 const sp<Client>& client, 3266 uint32_t sampleRate, 3267 audio_format_t format, 3268 uint32_t channelMask, 3269 int frameCount, 3270 const sp<IMemory>& sharedBuffer, 3271 int sessionId) 3272 : RefBase(), 3273 mThread(thread), 3274 mClient(client), 3275 mCblk(NULL), 3276 // mBuffer 3277 // mBufferEnd 3278 mFrameCount(0), 3279 mState(IDLE), 3280 mFormat(format), 3281 mStepServerFailed(false), 3282 mSessionId(sessionId) 3283 // mChannelCount 3284 // mChannelMask 3285{ 3286 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3287 3288 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3289 size_t size = sizeof(audio_track_cblk_t); 3290 uint8_t channelCount = popcount(channelMask); 3291 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3292 if (sharedBuffer == 0) { 3293 size += bufferSize; 3294 } 3295 3296 if (client != NULL) { 3297 mCblkMemory = client->heap()->allocate(size); 3298 if (mCblkMemory != 0) { 3299 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3300 if (mCblk != NULL) { // construct the shared structure in-place. 3301 new(mCblk) audio_track_cblk_t(); 3302 // clear all buffers 3303 mCblk->frameCount = frameCount; 3304 mCblk->sampleRate = sampleRate; 3305 mChannelCount = channelCount; 3306 mChannelMask = channelMask; 3307 if (sharedBuffer == 0) { 3308 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3309 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3310 // Force underrun condition to avoid false underrun callback until first data is 3311 // written to buffer (other flags are cleared) 3312 mCblk->flags = CBLK_UNDERRUN_ON; 3313 } else { 3314 mBuffer = sharedBuffer->pointer(); 3315 } 3316 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3317 } 3318 } else { 3319 ALOGE("not enough memory for AudioTrack size=%u", size); 3320 client->heap()->dump("AudioTrack"); 3321 return; 3322 } 3323 } else { 3324 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3325 // construct the shared structure in-place. 3326 new(mCblk) audio_track_cblk_t(); 3327 // clear all buffers 3328 mCblk->frameCount = frameCount; 3329 mCblk->sampleRate = sampleRate; 3330 mChannelCount = channelCount; 3331 mChannelMask = channelMask; 3332 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3333 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3334 // Force underrun condition to avoid false underrun callback until first data is 3335 // written to buffer (other flags are cleared) 3336 mCblk->flags = CBLK_UNDERRUN_ON; 3337 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3338 } 3339} 3340 3341AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3342{ 3343 if (mCblk != NULL) { 3344 if (mClient == 0) { 3345 delete mCblk; 3346 } else { 3347 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3348 } 3349 } 3350 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3351 if (mClient != 0) { 3352 // Client destructor must run with AudioFlinger mutex locked 3353 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3354 // If the client's reference count drops to zero, the associated destructor 3355 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3356 // relying on the automatic clear() at end of scope. 3357 mClient.clear(); 3358 } 3359} 3360 3361// AudioBufferProvider interface 3362// getNextBuffer() = 0; 3363// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3364void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3365{ 3366 buffer->raw = NULL; 3367 mFrameCount = buffer->frameCount; 3368 (void) step(); // ignore return value of step() 3369 buffer->frameCount = 0; 3370} 3371 3372bool AudioFlinger::ThreadBase::TrackBase::step() { 3373 bool result; 3374 audio_track_cblk_t* cblk = this->cblk(); 3375 3376 result = cblk->stepServer(mFrameCount); 3377 if (!result) { 3378 ALOGV("stepServer failed acquiring cblk mutex"); 3379 mStepServerFailed = true; 3380 } 3381 return result; 3382} 3383 3384void AudioFlinger::ThreadBase::TrackBase::reset() { 3385 audio_track_cblk_t* cblk = this->cblk(); 3386 3387 cblk->user = 0; 3388 cblk->server = 0; 3389 cblk->userBase = 0; 3390 cblk->serverBase = 0; 3391 mStepServerFailed = false; 3392 ALOGV("TrackBase::reset"); 3393} 3394 3395int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3396 return (int)mCblk->sampleRate; 3397} 3398 3399void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3400 audio_track_cblk_t* cblk = this->cblk(); 3401 size_t frameSize = cblk->frameSize; 3402 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3403 int8_t *bufferEnd = bufferStart + frames * frameSize; 3404 3405 // Check validity of returned pointer in case the track control block would have been corrupted. 3406 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3407 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3408 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3409 server %d, serverBase %d, user %d, userBase %d", 3410 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3411 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3412 return NULL; 3413 } 3414 3415 return bufferStart; 3416} 3417 3418// ---------------------------------------------------------------------------- 3419 3420// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3421AudioFlinger::PlaybackThread::Track::Track( 3422 PlaybackThread *thread, 3423 const sp<Client>& client, 3424 audio_stream_type_t streamType, 3425 uint32_t sampleRate, 3426 audio_format_t format, 3427 uint32_t channelMask, 3428 int frameCount, 3429 const sp<IMemory>& sharedBuffer, 3430 int sessionId) 3431 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3432 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3433 mAuxEffectId(0), mHasVolumeController(false) 3434{ 3435 if (mCblk != NULL) { 3436 if (thread != NULL) { 3437 mName = thread->getTrackName_l(); 3438 mMainBuffer = thread->mixBuffer(); 3439 } 3440 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3441 if (mName < 0) { 3442 ALOGE("no more track names available"); 3443 } 3444 mStreamType = streamType; 3445 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3446 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3447 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3448 } 3449} 3450 3451AudioFlinger::PlaybackThread::Track::~Track() 3452{ 3453 ALOGV("PlaybackThread::Track destructor"); 3454 sp<ThreadBase> thread = mThread.promote(); 3455 if (thread != 0) { 3456 Mutex::Autolock _l(thread->mLock); 3457 mState = TERMINATED; 3458 } 3459} 3460 3461void AudioFlinger::PlaybackThread::Track::destroy() 3462{ 3463 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3464 // by removing it from mTracks vector, so there is a risk that this Tracks's 3465 // destructor is called. As the destructor needs to lock mLock, 3466 // we must acquire a strong reference on this Track before locking mLock 3467 // here so that the destructor is called only when exiting this function. 3468 // On the other hand, as long as Track::destroy() is only called by 3469 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3470 // this Track with its member mTrack. 3471 sp<Track> keep(this); 3472 { // scope for mLock 3473 sp<ThreadBase> thread = mThread.promote(); 3474 if (thread != 0) { 3475 if (!isOutputTrack()) { 3476 if (mState == ACTIVE || mState == RESUMING) { 3477 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3478 3479 // to track the speaker usage 3480 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3481 } 3482 AudioSystem::releaseOutput(thread->id()); 3483 } 3484 Mutex::Autolock _l(thread->mLock); 3485 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3486 playbackThread->destroyTrack_l(this); 3487 } 3488 } 3489} 3490 3491void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3492{ 3493 uint32_t vlr = mCblk->getVolumeLR(); 3494 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", 3495 mName - AudioMixer::TRACK0, 3496 (mClient == 0) ? getpid_cached : mClient->pid(), 3497 mStreamType, 3498 mFormat, 3499 mChannelMask, 3500 mSessionId, 3501 mFrameCount, 3502 mState, 3503 mMute, 3504 mFillingUpStatus, 3505 mCblk->sampleRate, 3506 vlr & 0xFFFF, 3507 vlr >> 16, 3508 mCblk->server, 3509 mCblk->user, 3510 (int)mMainBuffer, 3511 (int)mAuxBuffer); 3512} 3513 3514// AudioBufferProvider interface 3515status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3516 AudioBufferProvider::Buffer* buffer, int64_t pts) 3517{ 3518 audio_track_cblk_t* cblk = this->cblk(); 3519 uint32_t framesReady; 3520 uint32_t framesReq = buffer->frameCount; 3521 3522 // Check if last stepServer failed, try to step now 3523 if (mStepServerFailed) { 3524 if (!step()) goto getNextBuffer_exit; 3525 ALOGV("stepServer recovered"); 3526 mStepServerFailed = false; 3527 } 3528 3529 framesReady = cblk->framesReady(); 3530 3531 if (CC_LIKELY(framesReady)) { 3532 uint32_t s = cblk->server; 3533 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3534 3535 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3536 if (framesReq > framesReady) { 3537 framesReq = framesReady; 3538 } 3539 if (s + framesReq > bufferEnd) { 3540 framesReq = bufferEnd - s; 3541 } 3542 3543 buffer->raw = getBuffer(s, framesReq); 3544 if (buffer->raw == NULL) goto getNextBuffer_exit; 3545 3546 buffer->frameCount = framesReq; 3547 return NO_ERROR; 3548 } 3549 3550getNextBuffer_exit: 3551 buffer->raw = NULL; 3552 buffer->frameCount = 0; 3553 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3554 return NOT_ENOUGH_DATA; 3555} 3556 3557uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3558 return mCblk->framesReady(); 3559} 3560 3561bool AudioFlinger::PlaybackThread::Track::isReady() const { 3562 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3563 3564 if (framesReady() >= mCblk->frameCount || 3565 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3566 mFillingUpStatus = FS_FILLED; 3567 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3568 return true; 3569 } 3570 return false; 3571} 3572 3573status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3574{ 3575 status_t status = NO_ERROR; 3576 ALOGV("start(%d), calling pid %d session %d tid %d", 3577 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3578 sp<ThreadBase> thread = mThread.promote(); 3579 if (thread != 0) { 3580 Mutex::Autolock _l(thread->mLock); 3581 track_state state = mState; 3582 // here the track could be either new, or restarted 3583 // in both cases "unstop" the track 3584 if (mState == PAUSED) { 3585 mState = TrackBase::RESUMING; 3586 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3587 } else { 3588 mState = TrackBase::ACTIVE; 3589 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3590 } 3591 3592 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3593 thread->mLock.unlock(); 3594 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3595 thread->mLock.lock(); 3596 3597 // to track the speaker usage 3598 if (status == NO_ERROR) { 3599 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3600 } 3601 } 3602 if (status == NO_ERROR) { 3603 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3604 playbackThread->addTrack_l(this); 3605 } else { 3606 mState = state; 3607 } 3608 } else { 3609 status = BAD_VALUE; 3610 } 3611 return status; 3612} 3613 3614void AudioFlinger::PlaybackThread::Track::stop() 3615{ 3616 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3617 sp<ThreadBase> thread = mThread.promote(); 3618 if (thread != 0) { 3619 Mutex::Autolock _l(thread->mLock); 3620 track_state state = mState; 3621 if (mState > STOPPED) { 3622 mState = STOPPED; 3623 // If the track is not active (PAUSED and buffers full), flush buffers 3624 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3625 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3626 reset(); 3627 } 3628 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3629 } 3630 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3631 thread->mLock.unlock(); 3632 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3633 thread->mLock.lock(); 3634 3635 // to track the speaker usage 3636 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3637 } 3638 } 3639} 3640 3641void AudioFlinger::PlaybackThread::Track::pause() 3642{ 3643 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3644 sp<ThreadBase> thread = mThread.promote(); 3645 if (thread != 0) { 3646 Mutex::Autolock _l(thread->mLock); 3647 if (mState == ACTIVE || mState == RESUMING) { 3648 mState = PAUSING; 3649 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3650 if (!isOutputTrack()) { 3651 thread->mLock.unlock(); 3652 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3653 thread->mLock.lock(); 3654 3655 // to track the speaker usage 3656 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3657 } 3658 } 3659 } 3660} 3661 3662void AudioFlinger::PlaybackThread::Track::flush() 3663{ 3664 ALOGV("flush(%d)", mName); 3665 sp<ThreadBase> thread = mThread.promote(); 3666 if (thread != 0) { 3667 Mutex::Autolock _l(thread->mLock); 3668 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3669 return; 3670 } 3671 // No point remaining in PAUSED state after a flush => go to 3672 // STOPPED state 3673 mState = STOPPED; 3674 3675 // do not reset the track if it is still in the process of being stopped or paused. 3676 // this will be done by prepareTracks_l() when the track is stopped. 3677 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3678 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3679 reset(); 3680 } 3681 } 3682} 3683 3684void AudioFlinger::PlaybackThread::Track::reset() 3685{ 3686 // Do not reset twice to avoid discarding data written just after a flush and before 3687 // the audioflinger thread detects the track is stopped. 3688 if (!mResetDone) { 3689 TrackBase::reset(); 3690 // Force underrun condition to avoid false underrun callback until first data is 3691 // written to buffer 3692 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3693 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3694 mFillingUpStatus = FS_FILLING; 3695 mResetDone = true; 3696 } 3697} 3698 3699void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3700{ 3701 mMute = muted; 3702} 3703 3704status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3705{ 3706 status_t status = DEAD_OBJECT; 3707 sp<ThreadBase> thread = mThread.promote(); 3708 if (thread != 0) { 3709 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3710 status = playbackThread->attachAuxEffect(this, EffectId); 3711 } 3712 return status; 3713} 3714 3715void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3716{ 3717 mAuxEffectId = EffectId; 3718 mAuxBuffer = buffer; 3719} 3720 3721// timed audio tracks 3722 3723sp<AudioFlinger::PlaybackThread::TimedTrack> 3724AudioFlinger::PlaybackThread::TimedTrack::create( 3725 PlaybackThread *thread, 3726 const sp<Client>& client, 3727 audio_stream_type_t streamType, 3728 uint32_t sampleRate, 3729 audio_format_t format, 3730 uint32_t channelMask, 3731 int frameCount, 3732 const sp<IMemory>& sharedBuffer, 3733 int sessionId) { 3734 if (!client->reserveTimedTrack()) 3735 return NULL; 3736 3737 sp<TimedTrack> track = new TimedTrack( 3738 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3739 sharedBuffer, sessionId); 3740 3741 if (track == NULL) { 3742 client->releaseTimedTrack(); 3743 return NULL; 3744 } 3745 3746 return track; 3747} 3748 3749AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3750 PlaybackThread *thread, 3751 const sp<Client>& client, 3752 audio_stream_type_t streamType, 3753 uint32_t sampleRate, 3754 audio_format_t format, 3755 uint32_t channelMask, 3756 int frameCount, 3757 const sp<IMemory>& sharedBuffer, 3758 int sessionId) 3759 : Track(thread, client, streamType, sampleRate, format, channelMask, 3760 frameCount, sharedBuffer, sessionId), 3761 mTimedSilenceBuffer(NULL), 3762 mTimedSilenceBufferSize(0), 3763 mTimedAudioOutputOnTime(false), 3764 mMediaTimeTransformValid(false) 3765{ 3766 LocalClock lc; 3767 mLocalTimeFreq = lc.getLocalFreq(); 3768 3769 mLocalTimeToSampleTransform.a_zero = 0; 3770 mLocalTimeToSampleTransform.b_zero = 0; 3771 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3772 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3773 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3774 &mLocalTimeToSampleTransform.a_to_b_denom); 3775} 3776 3777AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3778 mClient->releaseTimedTrack(); 3779 delete [] mTimedSilenceBuffer; 3780} 3781 3782status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3783 size_t size, sp<IMemory>* buffer) { 3784 3785 Mutex::Autolock _l(mTimedBufferQueueLock); 3786 3787 trimTimedBufferQueue_l(); 3788 3789 // lazily initialize the shared memory heap for timed buffers 3790 if (mTimedMemoryDealer == NULL) { 3791 const int kTimedBufferHeapSize = 512 << 10; 3792 3793 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3794 "AudioFlingerTimed"); 3795 if (mTimedMemoryDealer == NULL) 3796 return NO_MEMORY; 3797 } 3798 3799 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3800 if (newBuffer == NULL) { 3801 newBuffer = mTimedMemoryDealer->allocate(size); 3802 if (newBuffer == NULL) 3803 return NO_MEMORY; 3804 } 3805 3806 *buffer = newBuffer; 3807 return NO_ERROR; 3808} 3809 3810// caller must hold mTimedBufferQueueLock 3811void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3812 int64_t mediaTimeNow; 3813 { 3814 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3815 if (!mMediaTimeTransformValid) 3816 return; 3817 3818 int64_t targetTimeNow; 3819 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3820 ? mCCHelper.getCommonTime(&targetTimeNow) 3821 : mCCHelper.getLocalTime(&targetTimeNow); 3822 3823 if (OK != res) 3824 return; 3825 3826 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3827 &mediaTimeNow)) { 3828 return; 3829 } 3830 } 3831 3832 size_t trimIndex; 3833 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3834 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3835 break; 3836 } 3837 3838 if (trimIndex) { 3839 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3840 } 3841} 3842 3843status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3844 const sp<IMemory>& buffer, int64_t pts) { 3845 3846 { 3847 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3848 if (!mMediaTimeTransformValid) 3849 return INVALID_OPERATION; 3850 } 3851 3852 Mutex::Autolock _l(mTimedBufferQueueLock); 3853 3854 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3855 3856 return NO_ERROR; 3857} 3858 3859status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3860 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3861 3862 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3863 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3864 target); 3865 3866 if (!(target == TimedAudioTrack::LOCAL_TIME || 3867 target == TimedAudioTrack::COMMON_TIME)) { 3868 return BAD_VALUE; 3869 } 3870 3871 Mutex::Autolock lock(mMediaTimeTransformLock); 3872 mMediaTimeTransform = xform; 3873 mMediaTimeTransformTarget = target; 3874 mMediaTimeTransformValid = true; 3875 3876 return NO_ERROR; 3877} 3878 3879#define min(a, b) ((a) < (b) ? (a) : (b)) 3880 3881// implementation of getNextBuffer for tracks whose buffers have timestamps 3882status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3883 AudioBufferProvider::Buffer* buffer, int64_t pts) 3884{ 3885 if (pts == AudioBufferProvider::kInvalidPTS) { 3886 buffer->raw = 0; 3887 buffer->frameCount = 0; 3888 return INVALID_OPERATION; 3889 } 3890 3891 Mutex::Autolock _l(mTimedBufferQueueLock); 3892 3893 while (true) { 3894 3895 // if we have no timed buffers, then fail 3896 if (mTimedBufferQueue.isEmpty()) { 3897 buffer->raw = 0; 3898 buffer->frameCount = 0; 3899 return NOT_ENOUGH_DATA; 3900 } 3901 3902 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3903 3904 // calculate the PTS of the head of the timed buffer queue expressed in 3905 // local time 3906 int64_t headLocalPTS; 3907 { 3908 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3909 3910 assert(mMediaTimeTransformValid); 3911 3912 if (mMediaTimeTransform.a_to_b_denom == 0) { 3913 // the transform represents a pause, so yield silence 3914 timedYieldSilence(buffer->frameCount, buffer); 3915 return NO_ERROR; 3916 } 3917 3918 int64_t transformedPTS; 3919 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3920 &transformedPTS)) { 3921 // the transform failed. this shouldn't happen, but if it does 3922 // then just drop this buffer 3923 ALOGW("timedGetNextBuffer transform failed"); 3924 buffer->raw = 0; 3925 buffer->frameCount = 0; 3926 mTimedBufferQueue.removeAt(0); 3927 return NO_ERROR; 3928 } 3929 3930 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3931 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3932 &headLocalPTS)) { 3933 buffer->raw = 0; 3934 buffer->frameCount = 0; 3935 return INVALID_OPERATION; 3936 } 3937 } else { 3938 headLocalPTS = transformedPTS; 3939 } 3940 } 3941 3942 // adjust the head buffer's PTS to reflect the portion of the head buffer 3943 // that has already been consumed 3944 int64_t effectivePTS = headLocalPTS + 3945 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3946 3947 // Calculate the delta in samples between the head of the input buffer 3948 // queue and the start of the next output buffer that will be written. 3949 // If the transformation fails because of over or underflow, it means 3950 // that the sample's position in the output stream is so far out of 3951 // whack that it should just be dropped. 3952 int64_t sampleDelta; 3953 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3954 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3955 mTimedBufferQueue.removeAt(0); 3956 continue; 3957 } 3958 if (!mLocalTimeToSampleTransform.doForwardTransform( 3959 (effectivePTS - pts) << 32, &sampleDelta)) { 3960 ALOGV("*** too late during sample rate transform: dropped buffer"); 3961 mTimedBufferQueue.removeAt(0); 3962 continue; 3963 } 3964 3965 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3966 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3967 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3968 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3969 3970 // if the delta between the ideal placement for the next input sample and 3971 // the current output position is within this threshold, then we will 3972 // concatenate the next input samples to the previous output 3973 const int64_t kSampleContinuityThreshold = 3974 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3975 3976 // if this is the first buffer of audio that we're emitting from this track 3977 // then it should be almost exactly on time. 3978 const int64_t kSampleStartupThreshold = 1LL << 32; 3979 3980 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3981 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3982 // the next input is close enough to being on time, so concatenate it 3983 // with the last output 3984 timedYieldSamples(buffer); 3985 3986 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3987 return NO_ERROR; 3988 } else if (sampleDelta > 0) { 3989 // the gap between the current output position and the proper start of 3990 // the next input sample is too big, so fill it with silence 3991 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3992 3993 timedYieldSilence(framesUntilNextInput, buffer); 3994 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3995 return NO_ERROR; 3996 } else { 3997 // the next input sample is late 3998 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3999 size_t onTimeSamplePosition = 4000 head.position() + lateFrames * mCblk->frameSize; 4001 4002 if (onTimeSamplePosition > head.buffer()->size()) { 4003 // all the remaining samples in the head are too late, so 4004 // drop it and move on 4005 ALOGV("*** too late: dropped buffer"); 4006 mTimedBufferQueue.removeAt(0); 4007 continue; 4008 } else { 4009 // skip over the late samples 4010 head.setPosition(onTimeSamplePosition); 4011 4012 // yield the available samples 4013 timedYieldSamples(buffer); 4014 4015 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4016 return NO_ERROR; 4017 } 4018 } 4019 } 4020} 4021 4022// Yield samples from the timed buffer queue head up to the given output 4023// buffer's capacity. 4024// 4025// Caller must hold mTimedBufferQueueLock 4026void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4027 AudioBufferProvider::Buffer* buffer) { 4028 4029 const TimedBuffer& head = mTimedBufferQueue[0]; 4030 4031 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4032 head.position()); 4033 4034 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4035 mCblk->frameSize); 4036 size_t framesRequested = buffer->frameCount; 4037 buffer->frameCount = min(framesLeftInHead, framesRequested); 4038 4039 mTimedAudioOutputOnTime = true; 4040} 4041 4042// Yield samples of silence up to the given output buffer's capacity 4043// 4044// Caller must hold mTimedBufferQueueLock 4045void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4046 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4047 4048 // lazily allocate a buffer filled with silence 4049 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4050 delete [] mTimedSilenceBuffer; 4051 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4052 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4053 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4054 } 4055 4056 buffer->raw = mTimedSilenceBuffer; 4057 size_t framesRequested = buffer->frameCount; 4058 buffer->frameCount = min(numFrames, framesRequested); 4059 4060 mTimedAudioOutputOnTime = false; 4061} 4062 4063// AudioBufferProvider interface 4064void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4065 AudioBufferProvider::Buffer* buffer) { 4066 4067 Mutex::Autolock _l(mTimedBufferQueueLock); 4068 4069 // If the buffer which was just released is part of the buffer at the head 4070 // of the queue, be sure to update the amt of the buffer which has been 4071 // consumed. If the buffer being returned is not part of the head of the 4072 // queue, its either because the buffer is part of the silence buffer, or 4073 // because the head of the timed queue was trimmed after the mixer called 4074 // getNextBuffer but before the mixer called releaseBuffer. 4075 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4076 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4077 4078 void* start = head.buffer()->pointer(); 4079 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4080 4081 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4082 head.setPosition(head.position() + 4083 (buffer->frameCount * mCblk->frameSize)); 4084 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4085 mTimedBufferQueue.removeAt(0); 4086 } 4087 } 4088 } 4089 4090 buffer->raw = 0; 4091 buffer->frameCount = 0; 4092} 4093 4094uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4095 Mutex::Autolock _l(mTimedBufferQueueLock); 4096 4097 uint32_t frames = 0; 4098 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4099 const TimedBuffer& tb = mTimedBufferQueue[i]; 4100 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4101 } 4102 4103 return frames; 4104} 4105 4106AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4107 : mPTS(0), mPosition(0) {} 4108 4109AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4110 const sp<IMemory>& buffer, int64_t pts) 4111 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4112 4113// ---------------------------------------------------------------------------- 4114 4115// RecordTrack constructor must be called with AudioFlinger::mLock held 4116AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4117 RecordThread *thread, 4118 const sp<Client>& client, 4119 uint32_t sampleRate, 4120 audio_format_t format, 4121 uint32_t channelMask, 4122 int frameCount, 4123 int sessionId) 4124 : TrackBase(thread, client, sampleRate, format, 4125 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4126 mOverflow(false) 4127{ 4128 if (mCblk != NULL) { 4129 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4130 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4131 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4132 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4133 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4134 } else { 4135 mCblk->frameSize = sizeof(int8_t); 4136 } 4137 } 4138} 4139 4140AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4141{ 4142 sp<ThreadBase> thread = mThread.promote(); 4143 if (thread != 0) { 4144 AudioSystem::releaseInput(thread->id()); 4145 } 4146} 4147 4148// AudioBufferProvider interface 4149status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4150{ 4151 audio_track_cblk_t* cblk = this->cblk(); 4152 uint32_t framesAvail; 4153 uint32_t framesReq = buffer->frameCount; 4154 4155 // Check if last stepServer failed, try to step now 4156 if (mStepServerFailed) { 4157 if (!step()) goto getNextBuffer_exit; 4158 ALOGV("stepServer recovered"); 4159 mStepServerFailed = false; 4160 } 4161 4162 framesAvail = cblk->framesAvailable_l(); 4163 4164 if (CC_LIKELY(framesAvail)) { 4165 uint32_t s = cblk->server; 4166 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4167 4168 if (framesReq > framesAvail) { 4169 framesReq = framesAvail; 4170 } 4171 if (s + framesReq > bufferEnd) { 4172 framesReq = bufferEnd - s; 4173 } 4174 4175 buffer->raw = getBuffer(s, framesReq); 4176 if (buffer->raw == NULL) goto getNextBuffer_exit; 4177 4178 buffer->frameCount = framesReq; 4179 return NO_ERROR; 4180 } 4181 4182getNextBuffer_exit: 4183 buffer->raw = NULL; 4184 buffer->frameCount = 0; 4185 return NOT_ENOUGH_DATA; 4186} 4187 4188status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4189{ 4190 sp<ThreadBase> thread = mThread.promote(); 4191 if (thread != 0) { 4192 RecordThread *recordThread = (RecordThread *)thread.get(); 4193 return recordThread->start(this, tid); 4194 } else { 4195 return BAD_VALUE; 4196 } 4197} 4198 4199void AudioFlinger::RecordThread::RecordTrack::stop() 4200{ 4201 sp<ThreadBase> thread = mThread.promote(); 4202 if (thread != 0) { 4203 RecordThread *recordThread = (RecordThread *)thread.get(); 4204 recordThread->stop(this); 4205 TrackBase::reset(); 4206 // Force overerrun condition to avoid false overrun callback until first data is 4207 // read from buffer 4208 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4209 } 4210} 4211 4212void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4213{ 4214 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4215 (mClient == 0) ? getpid_cached : mClient->pid(), 4216 mFormat, 4217 mChannelMask, 4218 mSessionId, 4219 mFrameCount, 4220 mState, 4221 mCblk->sampleRate, 4222 mCblk->server, 4223 mCblk->user); 4224} 4225 4226 4227// ---------------------------------------------------------------------------- 4228 4229AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4230 PlaybackThread *playbackThread, 4231 DuplicatingThread *sourceThread, 4232 uint32_t sampleRate, 4233 audio_format_t format, 4234 uint32_t channelMask, 4235 int frameCount) 4236 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4237 mActive(false), mSourceThread(sourceThread) 4238{ 4239 4240 if (mCblk != NULL) { 4241 mCblk->flags |= CBLK_DIRECTION_OUT; 4242 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4243 mOutBuffer.frameCount = 0; 4244 playbackThread->mTracks.add(this); 4245 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4246 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4247 mCblk, mBuffer, mCblk->buffers, 4248 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4249 } else { 4250 ALOGW("Error creating output track on thread %p", playbackThread); 4251 } 4252} 4253 4254AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4255{ 4256 clearBufferQueue(); 4257} 4258 4259status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4260{ 4261 status_t status = Track::start(tid); 4262 if (status != NO_ERROR) { 4263 return status; 4264 } 4265 4266 mActive = true; 4267 mRetryCount = 127; 4268 return status; 4269} 4270 4271void AudioFlinger::PlaybackThread::OutputTrack::stop() 4272{ 4273 Track::stop(); 4274 clearBufferQueue(); 4275 mOutBuffer.frameCount = 0; 4276 mActive = false; 4277} 4278 4279bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4280{ 4281 Buffer *pInBuffer; 4282 Buffer inBuffer; 4283 uint32_t channelCount = mChannelCount; 4284 bool outputBufferFull = false; 4285 inBuffer.frameCount = frames; 4286 inBuffer.i16 = data; 4287 4288 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4289 4290 if (!mActive && frames != 0) { 4291 start(0); 4292 sp<ThreadBase> thread = mThread.promote(); 4293 if (thread != 0) { 4294 MixerThread *mixerThread = (MixerThread *)thread.get(); 4295 if (mCblk->frameCount > frames){ 4296 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4297 uint32_t startFrames = (mCblk->frameCount - frames); 4298 pInBuffer = new Buffer; 4299 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4300 pInBuffer->frameCount = startFrames; 4301 pInBuffer->i16 = pInBuffer->mBuffer; 4302 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4303 mBufferQueue.add(pInBuffer); 4304 } else { 4305 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4306 } 4307 } 4308 } 4309 } 4310 4311 while (waitTimeLeftMs) { 4312 // First write pending buffers, then new data 4313 if (mBufferQueue.size()) { 4314 pInBuffer = mBufferQueue.itemAt(0); 4315 } else { 4316 pInBuffer = &inBuffer; 4317 } 4318 4319 if (pInBuffer->frameCount == 0) { 4320 break; 4321 } 4322 4323 if (mOutBuffer.frameCount == 0) { 4324 mOutBuffer.frameCount = pInBuffer->frameCount; 4325 nsecs_t startTime = systemTime(); 4326 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4327 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4328 outputBufferFull = true; 4329 break; 4330 } 4331 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4332 if (waitTimeLeftMs >= waitTimeMs) { 4333 waitTimeLeftMs -= waitTimeMs; 4334 } else { 4335 waitTimeLeftMs = 0; 4336 } 4337 } 4338 4339 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4340 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4341 mCblk->stepUser(outFrames); 4342 pInBuffer->frameCount -= outFrames; 4343 pInBuffer->i16 += outFrames * channelCount; 4344 mOutBuffer.frameCount -= outFrames; 4345 mOutBuffer.i16 += outFrames * channelCount; 4346 4347 if (pInBuffer->frameCount == 0) { 4348 if (mBufferQueue.size()) { 4349 mBufferQueue.removeAt(0); 4350 delete [] pInBuffer->mBuffer; 4351 delete pInBuffer; 4352 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4353 } else { 4354 break; 4355 } 4356 } 4357 } 4358 4359 // If we could not write all frames, allocate a buffer and queue it for next time. 4360 if (inBuffer.frameCount) { 4361 sp<ThreadBase> thread = mThread.promote(); 4362 if (thread != 0 && !thread->standby()) { 4363 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4364 pInBuffer = new Buffer; 4365 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4366 pInBuffer->frameCount = inBuffer.frameCount; 4367 pInBuffer->i16 = pInBuffer->mBuffer; 4368 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4369 mBufferQueue.add(pInBuffer); 4370 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4371 } else { 4372 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4373 } 4374 } 4375 } 4376 4377 // Calling write() with a 0 length buffer, means that no more data will be written: 4378 // If no more buffers are pending, fill output track buffer to make sure it is started 4379 // by output mixer. 4380 if (frames == 0 && mBufferQueue.size() == 0) { 4381 if (mCblk->user < mCblk->frameCount) { 4382 frames = mCblk->frameCount - mCblk->user; 4383 pInBuffer = new Buffer; 4384 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4385 pInBuffer->frameCount = frames; 4386 pInBuffer->i16 = pInBuffer->mBuffer; 4387 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4388 mBufferQueue.add(pInBuffer); 4389 } else if (mActive) { 4390 stop(); 4391 } 4392 } 4393 4394 return outputBufferFull; 4395} 4396 4397status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4398{ 4399 int active; 4400 status_t result; 4401 audio_track_cblk_t* cblk = mCblk; 4402 uint32_t framesReq = buffer->frameCount; 4403 4404// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4405 buffer->frameCount = 0; 4406 4407 uint32_t framesAvail = cblk->framesAvailable(); 4408 4409 4410 if (framesAvail == 0) { 4411 Mutex::Autolock _l(cblk->lock); 4412 goto start_loop_here; 4413 while (framesAvail == 0) { 4414 active = mActive; 4415 if (CC_UNLIKELY(!active)) { 4416 ALOGV("Not active and NO_MORE_BUFFERS"); 4417 return NO_MORE_BUFFERS; 4418 } 4419 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4420 if (result != NO_ERROR) { 4421 return NO_MORE_BUFFERS; 4422 } 4423 // read the server count again 4424 start_loop_here: 4425 framesAvail = cblk->framesAvailable_l(); 4426 } 4427 } 4428 4429// if (framesAvail < framesReq) { 4430// return NO_MORE_BUFFERS; 4431// } 4432 4433 if (framesReq > framesAvail) { 4434 framesReq = framesAvail; 4435 } 4436 4437 uint32_t u = cblk->user; 4438 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4439 4440 if (u + framesReq > bufferEnd) { 4441 framesReq = bufferEnd - u; 4442 } 4443 4444 buffer->frameCount = framesReq; 4445 buffer->raw = (void *)cblk->buffer(u); 4446 return NO_ERROR; 4447} 4448 4449 4450void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4451{ 4452 size_t size = mBufferQueue.size(); 4453 4454 for (size_t i = 0; i < size; i++) { 4455 Buffer *pBuffer = mBufferQueue.itemAt(i); 4456 delete [] pBuffer->mBuffer; 4457 delete pBuffer; 4458 } 4459 mBufferQueue.clear(); 4460} 4461 4462// ---------------------------------------------------------------------------- 4463 4464AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4465 : RefBase(), 4466 mAudioFlinger(audioFlinger), 4467 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4468 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4469 mPid(pid), 4470 mTimedTrackCount(0) 4471{ 4472 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4473} 4474 4475// Client destructor must be called with AudioFlinger::mLock held 4476AudioFlinger::Client::~Client() 4477{ 4478 mAudioFlinger->removeClient_l(mPid); 4479} 4480 4481sp<MemoryDealer> AudioFlinger::Client::heap() const 4482{ 4483 return mMemoryDealer; 4484} 4485 4486// Reserve one of the limited slots for a timed audio track associated 4487// with this client 4488bool AudioFlinger::Client::reserveTimedTrack() 4489{ 4490 const int kMaxTimedTracksPerClient = 4; 4491 4492 Mutex::Autolock _l(mTimedTrackLock); 4493 4494 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4495 ALOGW("can not create timed track - pid %d has exceeded the limit", 4496 mPid); 4497 return false; 4498 } 4499 4500 mTimedTrackCount++; 4501 return true; 4502} 4503 4504// Release a slot for a timed audio track 4505void AudioFlinger::Client::releaseTimedTrack() 4506{ 4507 Mutex::Autolock _l(mTimedTrackLock); 4508 mTimedTrackCount--; 4509} 4510 4511// ---------------------------------------------------------------------------- 4512 4513AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4514 const sp<IAudioFlingerClient>& client, 4515 pid_t pid) 4516 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4517{ 4518} 4519 4520AudioFlinger::NotificationClient::~NotificationClient() 4521{ 4522} 4523 4524void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4525{ 4526 sp<NotificationClient> keep(this); 4527 mAudioFlinger->removeNotificationClient(mPid); 4528} 4529 4530// ---------------------------------------------------------------------------- 4531 4532AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4533 : BnAudioTrack(), 4534 mTrack(track) 4535{ 4536} 4537 4538AudioFlinger::TrackHandle::~TrackHandle() { 4539 // just stop the track on deletion, associated resources 4540 // will be freed from the main thread once all pending buffers have 4541 // been played. Unless it's not in the active track list, in which 4542 // case we free everything now... 4543 mTrack->destroy(); 4544} 4545 4546sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4547 return mTrack->getCblk(); 4548} 4549 4550status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4551 return mTrack->start(tid); 4552} 4553 4554void AudioFlinger::TrackHandle::stop() { 4555 mTrack->stop(); 4556} 4557 4558void AudioFlinger::TrackHandle::flush() { 4559 mTrack->flush(); 4560} 4561 4562void AudioFlinger::TrackHandle::mute(bool e) { 4563 mTrack->mute(e); 4564} 4565 4566void AudioFlinger::TrackHandle::pause() { 4567 mTrack->pause(); 4568} 4569 4570status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4571{ 4572 return mTrack->attachAuxEffect(EffectId); 4573} 4574 4575status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4576 sp<IMemory>* buffer) { 4577 if (!mTrack->isTimedTrack()) 4578 return INVALID_OPERATION; 4579 4580 PlaybackThread::TimedTrack* tt = 4581 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4582 return tt->allocateTimedBuffer(size, buffer); 4583} 4584 4585status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4586 int64_t pts) { 4587 if (!mTrack->isTimedTrack()) 4588 return INVALID_OPERATION; 4589 4590 PlaybackThread::TimedTrack* tt = 4591 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4592 return tt->queueTimedBuffer(buffer, pts); 4593} 4594 4595status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4596 const LinearTransform& xform, int target) { 4597 4598 if (!mTrack->isTimedTrack()) 4599 return INVALID_OPERATION; 4600 4601 PlaybackThread::TimedTrack* tt = 4602 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4603 return tt->setMediaTimeTransform( 4604 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4605} 4606 4607status_t AudioFlinger::TrackHandle::onTransact( 4608 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4609{ 4610 return BnAudioTrack::onTransact(code, data, reply, flags); 4611} 4612 4613// ---------------------------------------------------------------------------- 4614 4615sp<IAudioRecord> AudioFlinger::openRecord( 4616 pid_t pid, 4617 audio_io_handle_t input, 4618 uint32_t sampleRate, 4619 audio_format_t format, 4620 uint32_t channelMask, 4621 int frameCount, 4622 // FIXME dead, remove from IAudioFlinger 4623 uint32_t flags, 4624 int *sessionId, 4625 status_t *status) 4626{ 4627 sp<RecordThread::RecordTrack> recordTrack; 4628 sp<RecordHandle> recordHandle; 4629 sp<Client> client; 4630 status_t lStatus; 4631 RecordThread *thread; 4632 size_t inFrameCount; 4633 int lSessionId; 4634 4635 // check calling permissions 4636 if (!recordingAllowed()) { 4637 lStatus = PERMISSION_DENIED; 4638 goto Exit; 4639 } 4640 4641 // add client to list 4642 { // scope for mLock 4643 Mutex::Autolock _l(mLock); 4644 thread = checkRecordThread_l(input); 4645 if (thread == NULL) { 4646 lStatus = BAD_VALUE; 4647 goto Exit; 4648 } 4649 4650 client = registerPid_l(pid); 4651 4652 // If no audio session id is provided, create one here 4653 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4654 lSessionId = *sessionId; 4655 } else { 4656 lSessionId = nextUniqueId(); 4657 if (sessionId != NULL) { 4658 *sessionId = lSessionId; 4659 } 4660 } 4661 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4662 recordTrack = thread->createRecordTrack_l(client, 4663 sampleRate, 4664 format, 4665 channelMask, 4666 frameCount, 4667 lSessionId, 4668 &lStatus); 4669 } 4670 if (lStatus != NO_ERROR) { 4671 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4672 // destructor is called by the TrackBase destructor with mLock held 4673 client.clear(); 4674 recordTrack.clear(); 4675 goto Exit; 4676 } 4677 4678 // return to handle to client 4679 recordHandle = new RecordHandle(recordTrack); 4680 lStatus = NO_ERROR; 4681 4682Exit: 4683 if (status) { 4684 *status = lStatus; 4685 } 4686 return recordHandle; 4687} 4688 4689// ---------------------------------------------------------------------------- 4690 4691AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4692 : BnAudioRecord(), 4693 mRecordTrack(recordTrack) 4694{ 4695} 4696 4697AudioFlinger::RecordHandle::~RecordHandle() { 4698 stop(); 4699} 4700 4701sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4702 return mRecordTrack->getCblk(); 4703} 4704 4705status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4706 ALOGV("RecordHandle::start()"); 4707 return mRecordTrack->start(tid); 4708} 4709 4710void AudioFlinger::RecordHandle::stop() { 4711 ALOGV("RecordHandle::stop()"); 4712 mRecordTrack->stop(); 4713} 4714 4715status_t AudioFlinger::RecordHandle::onTransact( 4716 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4717{ 4718 return BnAudioRecord::onTransact(code, data, reply, flags); 4719} 4720 4721// ---------------------------------------------------------------------------- 4722 4723AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4724 AudioStreamIn *input, 4725 uint32_t sampleRate, 4726 uint32_t channels, 4727 audio_io_handle_t id, 4728 uint32_t device) : 4729 ThreadBase(audioFlinger, id, device, RECORD), 4730 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4731 // mRsmpInIndex and mInputBytes set by readInputParameters() 4732 mReqChannelCount(popcount(channels)), 4733 mReqSampleRate(sampleRate) 4734 // mBytesRead is only meaningful while active, and so is cleared in start() 4735 // (but might be better to also clear here for dump?) 4736{ 4737 snprintf(mName, kNameLength, "AudioIn_%X", id); 4738 4739 readInputParameters(); 4740} 4741 4742 4743AudioFlinger::RecordThread::~RecordThread() 4744{ 4745 delete[] mRsmpInBuffer; 4746 delete mResampler; 4747 delete[] mRsmpOutBuffer; 4748} 4749 4750void AudioFlinger::RecordThread::onFirstRef() 4751{ 4752 run(mName, PRIORITY_URGENT_AUDIO); 4753} 4754 4755status_t AudioFlinger::RecordThread::readyToRun() 4756{ 4757 status_t status = initCheck(); 4758 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4759 return status; 4760} 4761 4762bool AudioFlinger::RecordThread::threadLoop() 4763{ 4764 AudioBufferProvider::Buffer buffer; 4765 sp<RecordTrack> activeTrack; 4766 Vector< sp<EffectChain> > effectChains; 4767 4768 nsecs_t lastWarning = 0; 4769 4770 acquireWakeLock(); 4771 4772 // start recording 4773 while (!exitPending()) { 4774 4775 processConfigEvents(); 4776 4777 { // scope for mLock 4778 Mutex::Autolock _l(mLock); 4779 checkForNewParameters_l(); 4780 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4781 if (!mStandby) { 4782 mInput->stream->common.standby(&mInput->stream->common); 4783 mStandby = true; 4784 } 4785 4786 if (exitPending()) break; 4787 4788 releaseWakeLock_l(); 4789 ALOGV("RecordThread: loop stopping"); 4790 // go to sleep 4791 mWaitWorkCV.wait(mLock); 4792 ALOGV("RecordThread: loop starting"); 4793 acquireWakeLock_l(); 4794 continue; 4795 } 4796 if (mActiveTrack != 0) { 4797 if (mActiveTrack->mState == TrackBase::PAUSING) { 4798 if (!mStandby) { 4799 mInput->stream->common.standby(&mInput->stream->common); 4800 mStandby = true; 4801 } 4802 mActiveTrack.clear(); 4803 mStartStopCond.broadcast(); 4804 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4805 if (mReqChannelCount != mActiveTrack->channelCount()) { 4806 mActiveTrack.clear(); 4807 mStartStopCond.broadcast(); 4808 } else if (mBytesRead != 0) { 4809 // record start succeeds only if first read from audio input 4810 // succeeds 4811 if (mBytesRead > 0) { 4812 mActiveTrack->mState = TrackBase::ACTIVE; 4813 } else { 4814 mActiveTrack.clear(); 4815 } 4816 mStartStopCond.broadcast(); 4817 } 4818 mStandby = false; 4819 } 4820 } 4821 lockEffectChains_l(effectChains); 4822 } 4823 4824 if (mActiveTrack != 0) { 4825 if (mActiveTrack->mState != TrackBase::ACTIVE && 4826 mActiveTrack->mState != TrackBase::RESUMING) { 4827 unlockEffectChains(effectChains); 4828 usleep(kRecordThreadSleepUs); 4829 continue; 4830 } 4831 for (size_t i = 0; i < effectChains.size(); i ++) { 4832 effectChains[i]->process_l(); 4833 } 4834 4835 buffer.frameCount = mFrameCount; 4836 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4837 size_t framesOut = buffer.frameCount; 4838 if (mResampler == NULL) { 4839 // no resampling 4840 while (framesOut) { 4841 size_t framesIn = mFrameCount - mRsmpInIndex; 4842 if (framesIn) { 4843 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4844 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4845 if (framesIn > framesOut) 4846 framesIn = framesOut; 4847 mRsmpInIndex += framesIn; 4848 framesOut -= framesIn; 4849 if ((int)mChannelCount == mReqChannelCount || 4850 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4851 memcpy(dst, src, framesIn * mFrameSize); 4852 } else { 4853 int16_t *src16 = (int16_t *)src; 4854 int16_t *dst16 = (int16_t *)dst; 4855 if (mChannelCount == 1) { 4856 while (framesIn--) { 4857 *dst16++ = *src16; 4858 *dst16++ = *src16++; 4859 } 4860 } else { 4861 while (framesIn--) { 4862 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4863 src16 += 2; 4864 } 4865 } 4866 } 4867 } 4868 if (framesOut && mFrameCount == mRsmpInIndex) { 4869 if (framesOut == mFrameCount && 4870 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4871 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4872 framesOut = 0; 4873 } else { 4874 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4875 mRsmpInIndex = 0; 4876 } 4877 if (mBytesRead < 0) { 4878 ALOGE("Error reading audio input"); 4879 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4880 // Force input into standby so that it tries to 4881 // recover at next read attempt 4882 mInput->stream->common.standby(&mInput->stream->common); 4883 usleep(kRecordThreadSleepUs); 4884 } 4885 mRsmpInIndex = mFrameCount; 4886 framesOut = 0; 4887 buffer.frameCount = 0; 4888 } 4889 } 4890 } 4891 } else { 4892 // resampling 4893 4894 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4895 // alter output frame count as if we were expecting stereo samples 4896 if (mChannelCount == 1 && mReqChannelCount == 1) { 4897 framesOut >>= 1; 4898 } 4899 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4900 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4901 // are 32 bit aligned which should be always true. 4902 if (mChannelCount == 2 && mReqChannelCount == 1) { 4903 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4904 // the resampler always outputs stereo samples: do post stereo to mono conversion 4905 int16_t *src = (int16_t *)mRsmpOutBuffer; 4906 int16_t *dst = buffer.i16; 4907 while (framesOut--) { 4908 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4909 src += 2; 4910 } 4911 } else { 4912 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4913 } 4914 4915 } 4916 mActiveTrack->releaseBuffer(&buffer); 4917 mActiveTrack->overflow(); 4918 } 4919 // client isn't retrieving buffers fast enough 4920 else { 4921 if (!mActiveTrack->setOverflow()) { 4922 nsecs_t now = systemTime(); 4923 if ((now - lastWarning) > kWarningThrottleNs) { 4924 ALOGW("RecordThread: buffer overflow"); 4925 lastWarning = now; 4926 } 4927 } 4928 // Release the processor for a while before asking for a new buffer. 4929 // This will give the application more chance to read from the buffer and 4930 // clear the overflow. 4931 usleep(kRecordThreadSleepUs); 4932 } 4933 } 4934 // enable changes in effect chain 4935 unlockEffectChains(effectChains); 4936 effectChains.clear(); 4937 } 4938 4939 if (!mStandby) { 4940 mInput->stream->common.standby(&mInput->stream->common); 4941 } 4942 mActiveTrack.clear(); 4943 4944 mStartStopCond.broadcast(); 4945 4946 releaseWakeLock(); 4947 4948 ALOGV("RecordThread %p exiting", this); 4949 return false; 4950} 4951 4952 4953sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4954 const sp<AudioFlinger::Client>& client, 4955 uint32_t sampleRate, 4956 audio_format_t format, 4957 int channelMask, 4958 int frameCount, 4959 int sessionId, 4960 status_t *status) 4961{ 4962 sp<RecordTrack> track; 4963 status_t lStatus; 4964 4965 lStatus = initCheck(); 4966 if (lStatus != NO_ERROR) { 4967 ALOGE("Audio driver not initialized."); 4968 goto Exit; 4969 } 4970 4971 { // scope for mLock 4972 Mutex::Autolock _l(mLock); 4973 4974 track = new RecordTrack(this, client, sampleRate, 4975 format, channelMask, frameCount, sessionId); 4976 4977 if (track->getCblk() == 0) { 4978 lStatus = NO_MEMORY; 4979 goto Exit; 4980 } 4981 4982 mTrack = track.get(); 4983 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4984 bool suspend = audio_is_bluetooth_sco_device( 4985 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4986 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4987 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4988 } 4989 lStatus = NO_ERROR; 4990 4991Exit: 4992 if (status) { 4993 *status = lStatus; 4994 } 4995 return track; 4996} 4997 4998status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4999{ 5000 ALOGV("RecordThread::start tid=%d", tid); 5001 sp <ThreadBase> strongMe = this; 5002 status_t status = NO_ERROR; 5003 { 5004 AutoMutex lock(mLock); 5005 if (mActiveTrack != 0) { 5006 if (recordTrack != mActiveTrack.get()) { 5007 status = -EBUSY; 5008 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5009 mActiveTrack->mState = TrackBase::ACTIVE; 5010 } 5011 return status; 5012 } 5013 5014 recordTrack->mState = TrackBase::IDLE; 5015 mActiveTrack = recordTrack; 5016 mLock.unlock(); 5017 status_t status = AudioSystem::startInput(mId); 5018 mLock.lock(); 5019 if (status != NO_ERROR) { 5020 mActiveTrack.clear(); 5021 return status; 5022 } 5023 mRsmpInIndex = mFrameCount; 5024 mBytesRead = 0; 5025 if (mResampler != NULL) { 5026 mResampler->reset(); 5027 } 5028 mActiveTrack->mState = TrackBase::RESUMING; 5029 // signal thread to start 5030 ALOGV("Signal record thread"); 5031 mWaitWorkCV.signal(); 5032 // do not wait for mStartStopCond if exiting 5033 if (exitPending()) { 5034 mActiveTrack.clear(); 5035 status = INVALID_OPERATION; 5036 goto startError; 5037 } 5038 mStartStopCond.wait(mLock); 5039 if (mActiveTrack == 0) { 5040 ALOGV("Record failed to start"); 5041 status = BAD_VALUE; 5042 goto startError; 5043 } 5044 ALOGV("Record started OK"); 5045 return status; 5046 } 5047startError: 5048 AudioSystem::stopInput(mId); 5049 return status; 5050} 5051 5052void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5053 ALOGV("RecordThread::stop"); 5054 sp <ThreadBase> strongMe = this; 5055 { 5056 AutoMutex lock(mLock); 5057 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5058 mActiveTrack->mState = TrackBase::PAUSING; 5059 // do not wait for mStartStopCond if exiting 5060 if (exitPending()) { 5061 return; 5062 } 5063 mStartStopCond.wait(mLock); 5064 // if we have been restarted, recordTrack == mActiveTrack.get() here 5065 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5066 mLock.unlock(); 5067 AudioSystem::stopInput(mId); 5068 mLock.lock(); 5069 ALOGV("Record stopped OK"); 5070 } 5071 } 5072 } 5073} 5074 5075status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5076{ 5077 const size_t SIZE = 256; 5078 char buffer[SIZE]; 5079 String8 result; 5080 5081 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5082 result.append(buffer); 5083 5084 if (mActiveTrack != 0) { 5085 result.append("Active Track:\n"); 5086 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5087 mActiveTrack->dump(buffer, SIZE); 5088 result.append(buffer); 5089 5090 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5091 result.append(buffer); 5092 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5093 result.append(buffer); 5094 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5095 result.append(buffer); 5096 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5097 result.append(buffer); 5098 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5099 result.append(buffer); 5100 5101 5102 } else { 5103 result.append("No record client\n"); 5104 } 5105 write(fd, result.string(), result.size()); 5106 5107 dumpBase(fd, args); 5108 dumpEffectChains(fd, args); 5109 5110 return NO_ERROR; 5111} 5112 5113// AudioBufferProvider interface 5114status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5115{ 5116 size_t framesReq = buffer->frameCount; 5117 size_t framesReady = mFrameCount - mRsmpInIndex; 5118 int channelCount; 5119 5120 if (framesReady == 0) { 5121 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5122 if (mBytesRead < 0) { 5123 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5124 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5125 // Force input into standby so that it tries to 5126 // recover at next read attempt 5127 mInput->stream->common.standby(&mInput->stream->common); 5128 usleep(kRecordThreadSleepUs); 5129 } 5130 buffer->raw = NULL; 5131 buffer->frameCount = 0; 5132 return NOT_ENOUGH_DATA; 5133 } 5134 mRsmpInIndex = 0; 5135 framesReady = mFrameCount; 5136 } 5137 5138 if (framesReq > framesReady) { 5139 framesReq = framesReady; 5140 } 5141 5142 if (mChannelCount == 1 && mReqChannelCount == 2) { 5143 channelCount = 1; 5144 } else { 5145 channelCount = 2; 5146 } 5147 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5148 buffer->frameCount = framesReq; 5149 return NO_ERROR; 5150} 5151 5152// AudioBufferProvider interface 5153void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5154{ 5155 mRsmpInIndex += buffer->frameCount; 5156 buffer->frameCount = 0; 5157} 5158 5159bool AudioFlinger::RecordThread::checkForNewParameters_l() 5160{ 5161 bool reconfig = false; 5162 5163 while (!mNewParameters.isEmpty()) { 5164 status_t status = NO_ERROR; 5165 String8 keyValuePair = mNewParameters[0]; 5166 AudioParameter param = AudioParameter(keyValuePair); 5167 int value; 5168 audio_format_t reqFormat = mFormat; 5169 int reqSamplingRate = mReqSampleRate; 5170 int reqChannelCount = mReqChannelCount; 5171 5172 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5173 reqSamplingRate = value; 5174 reconfig = true; 5175 } 5176 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5177 reqFormat = (audio_format_t) value; 5178 reconfig = true; 5179 } 5180 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5181 reqChannelCount = popcount(value); 5182 reconfig = true; 5183 } 5184 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5185 // do not accept frame count changes if tracks are open as the track buffer 5186 // size depends on frame count and correct behavior would not be guaranteed 5187 // if frame count is changed after track creation 5188 if (mActiveTrack != 0) { 5189 status = INVALID_OPERATION; 5190 } else { 5191 reconfig = true; 5192 } 5193 } 5194 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5195 // forward device change to effects that have requested to be 5196 // aware of attached audio device. 5197 for (size_t i = 0; i < mEffectChains.size(); i++) { 5198 mEffectChains[i]->setDevice_l(value); 5199 } 5200 // store input device and output device but do not forward output device to audio HAL. 5201 // Note that status is ignored by the caller for output device 5202 // (see AudioFlinger::setParameters() 5203 if (value & AUDIO_DEVICE_OUT_ALL) { 5204 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5205 status = BAD_VALUE; 5206 } else { 5207 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5208 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5209 if (mTrack != NULL) { 5210 bool suspend = audio_is_bluetooth_sco_device( 5211 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5212 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5213 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5214 } 5215 } 5216 mDevice |= (uint32_t)value; 5217 } 5218 if (status == NO_ERROR) { 5219 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5220 if (status == INVALID_OPERATION) { 5221 mInput->stream->common.standby(&mInput->stream->common); 5222 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5223 } 5224 if (reconfig) { 5225 if (status == BAD_VALUE && 5226 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5227 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5228 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5229 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5230 (reqChannelCount < 3)) { 5231 status = NO_ERROR; 5232 } 5233 if (status == NO_ERROR) { 5234 readInputParameters(); 5235 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5236 } 5237 } 5238 } 5239 5240 mNewParameters.removeAt(0); 5241 5242 mParamStatus = status; 5243 mParamCond.signal(); 5244 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5245 // already timed out waiting for the status and will never signal the condition. 5246 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5247 } 5248 return reconfig; 5249} 5250 5251String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5252{ 5253 char *s; 5254 String8 out_s8 = String8(); 5255 5256 Mutex::Autolock _l(mLock); 5257 if (initCheck() != NO_ERROR) { 5258 return out_s8; 5259 } 5260 5261 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5262 out_s8 = String8(s); 5263 free(s); 5264 return out_s8; 5265} 5266 5267void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5268 AudioSystem::OutputDescriptor desc; 5269 void *param2 = NULL; 5270 5271 switch (event) { 5272 case AudioSystem::INPUT_OPENED: 5273 case AudioSystem::INPUT_CONFIG_CHANGED: 5274 desc.channels = mChannelMask; 5275 desc.samplingRate = mSampleRate; 5276 desc.format = mFormat; 5277 desc.frameCount = mFrameCount; 5278 desc.latency = 0; 5279 param2 = &desc; 5280 break; 5281 5282 case AudioSystem::INPUT_CLOSED: 5283 default: 5284 break; 5285 } 5286 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5287} 5288 5289void AudioFlinger::RecordThread::readInputParameters() 5290{ 5291 delete mRsmpInBuffer; 5292 // mRsmpInBuffer is always assigned a new[] below 5293 delete mRsmpOutBuffer; 5294 mRsmpOutBuffer = NULL; 5295 delete mResampler; 5296 mResampler = NULL; 5297 5298 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5299 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5300 mChannelCount = (uint16_t)popcount(mChannelMask); 5301 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5302 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5303 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5304 mFrameCount = mInputBytes / mFrameSize; 5305 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5306 5307 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5308 { 5309 int channelCount; 5310 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5311 // stereo to mono post process as the resampler always outputs stereo. 5312 if (mChannelCount == 1 && mReqChannelCount == 2) { 5313 channelCount = 1; 5314 } else { 5315 channelCount = 2; 5316 } 5317 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5318 mResampler->setSampleRate(mSampleRate); 5319 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5320 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5321 5322 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5323 if (mChannelCount == 1 && mReqChannelCount == 1) { 5324 mFrameCount >>= 1; 5325 } 5326 5327 } 5328 mRsmpInIndex = mFrameCount; 5329} 5330 5331unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5332{ 5333 Mutex::Autolock _l(mLock); 5334 if (initCheck() != NO_ERROR) { 5335 return 0; 5336 } 5337 5338 return mInput->stream->get_input_frames_lost(mInput->stream); 5339} 5340 5341uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5342{ 5343 Mutex::Autolock _l(mLock); 5344 uint32_t result = 0; 5345 if (getEffectChain_l(sessionId) != 0) { 5346 result = EFFECT_SESSION; 5347 } 5348 5349 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5350 result |= TRACK_SESSION; 5351 } 5352 5353 return result; 5354} 5355 5356AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5357{ 5358 Mutex::Autolock _l(mLock); 5359 return mTrack; 5360} 5361 5362AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5363{ 5364 Mutex::Autolock _l(mLock); 5365 return mInput; 5366} 5367 5368AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5369{ 5370 Mutex::Autolock _l(mLock); 5371 AudioStreamIn *input = mInput; 5372 mInput = NULL; 5373 return input; 5374} 5375 5376// this method must always be called either with ThreadBase mLock held or inside the thread loop 5377audio_stream_t* AudioFlinger::RecordThread::stream() 5378{ 5379 if (mInput == NULL) { 5380 return NULL; 5381 } 5382 return &mInput->stream->common; 5383} 5384 5385 5386// ---------------------------------------------------------------------------- 5387 5388audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5389 uint32_t *pSamplingRate, 5390 audio_format_t *pFormat, 5391 uint32_t *pChannels, 5392 uint32_t *pLatencyMs, 5393 uint32_t flags) 5394{ 5395 status_t status; 5396 PlaybackThread *thread = NULL; 5397 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5398 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5399 uint32_t channels = pChannels ? *pChannels : 0; 5400 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5401 audio_stream_out_t *outStream; 5402 audio_hw_device_t *outHwDev; 5403 5404 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5405 pDevices ? *pDevices : 0, 5406 samplingRate, 5407 format, 5408 channels, 5409 flags); 5410 5411 if (pDevices == NULL || *pDevices == 0) { 5412 return 0; 5413 } 5414 5415 Mutex::Autolock _l(mLock); 5416 5417 outHwDev = findSuitableHwDev_l(*pDevices); 5418 if (outHwDev == NULL) 5419 return 0; 5420 5421 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5422 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5423 &channels, &samplingRate, &outStream); 5424 mHardwareStatus = AUDIO_HW_IDLE; 5425 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5426 outStream, 5427 samplingRate, 5428 format, 5429 channels, 5430 status); 5431 5432 if (outStream != NULL) { 5433 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5434 audio_io_handle_t id = nextUniqueId(); 5435 5436 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5437 (format != AUDIO_FORMAT_PCM_16_BIT) || 5438 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5439 thread = new DirectOutputThread(this, output, id, *pDevices); 5440 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5441 } else { 5442 thread = new MixerThread(this, output, id, *pDevices); 5443 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5444 } 5445 mPlaybackThreads.add(id, thread); 5446 5447 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5448 if (pFormat != NULL) *pFormat = format; 5449 if (pChannels != NULL) *pChannels = channels; 5450 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5451 5452 // notify client processes of the new output creation 5453 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5454 return id; 5455 } 5456 5457 return 0; 5458} 5459 5460audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5461 audio_io_handle_t output2) 5462{ 5463 Mutex::Autolock _l(mLock); 5464 MixerThread *thread1 = checkMixerThread_l(output1); 5465 MixerThread *thread2 = checkMixerThread_l(output2); 5466 5467 if (thread1 == NULL || thread2 == NULL) { 5468 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5469 return 0; 5470 } 5471 5472 audio_io_handle_t id = nextUniqueId(); 5473 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5474 thread->addOutputTrack(thread2); 5475 mPlaybackThreads.add(id, thread); 5476 // notify client processes of the new output creation 5477 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5478 return id; 5479} 5480 5481status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5482{ 5483 // keep strong reference on the playback thread so that 5484 // it is not destroyed while exit() is executed 5485 sp <PlaybackThread> thread; 5486 { 5487 Mutex::Autolock _l(mLock); 5488 thread = checkPlaybackThread_l(output); 5489 if (thread == NULL) { 5490 return BAD_VALUE; 5491 } 5492 5493 ALOGV("closeOutput() %d", output); 5494 5495 if (thread->type() == ThreadBase::MIXER) { 5496 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5497 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5498 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5499 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5500 } 5501 } 5502 } 5503 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5504 mPlaybackThreads.removeItem(output); 5505 } 5506 thread->exit(); 5507 // The thread entity (active unit of execution) is no longer running here, 5508 // but the ThreadBase container still exists. 5509 5510 if (thread->type() != ThreadBase::DUPLICATING) { 5511 AudioStreamOut *out = thread->clearOutput(); 5512 assert(out != NULL); 5513 // from now on thread->mOutput is NULL 5514 out->hwDev->close_output_stream(out->hwDev, out->stream); 5515 delete out; 5516 } 5517 return NO_ERROR; 5518} 5519 5520status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5521{ 5522 Mutex::Autolock _l(mLock); 5523 PlaybackThread *thread = checkPlaybackThread_l(output); 5524 5525 if (thread == NULL) { 5526 return BAD_VALUE; 5527 } 5528 5529 ALOGV("suspendOutput() %d", output); 5530 thread->suspend(); 5531 5532 return NO_ERROR; 5533} 5534 5535status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5536{ 5537 Mutex::Autolock _l(mLock); 5538 PlaybackThread *thread = checkPlaybackThread_l(output); 5539 5540 if (thread == NULL) { 5541 return BAD_VALUE; 5542 } 5543 5544 ALOGV("restoreOutput() %d", output); 5545 5546 thread->restore(); 5547 5548 return NO_ERROR; 5549} 5550 5551audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5552 uint32_t *pSamplingRate, 5553 audio_format_t *pFormat, 5554 uint32_t *pChannels, 5555 audio_in_acoustics_t acoustics) 5556{ 5557 status_t status; 5558 RecordThread *thread = NULL; 5559 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5560 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5561 uint32_t channels = pChannels ? *pChannels : 0; 5562 uint32_t reqSamplingRate = samplingRate; 5563 audio_format_t reqFormat = format; 5564 uint32_t reqChannels = channels; 5565 audio_stream_in_t *inStream; 5566 audio_hw_device_t *inHwDev; 5567 5568 if (pDevices == NULL || *pDevices == 0) { 5569 return 0; 5570 } 5571 5572 Mutex::Autolock _l(mLock); 5573 5574 inHwDev = findSuitableHwDev_l(*pDevices); 5575 if (inHwDev == NULL) 5576 return 0; 5577 5578 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5579 &channels, &samplingRate, 5580 acoustics, 5581 &inStream); 5582 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5583 inStream, 5584 samplingRate, 5585 format, 5586 channels, 5587 acoustics, 5588 status); 5589 5590 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5591 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5592 // or stereo to mono conversions on 16 bit PCM inputs. 5593 if (inStream == NULL && status == BAD_VALUE && 5594 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5595 (samplingRate <= 2 * reqSamplingRate) && 5596 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5597 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5598 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5599 &channels, &samplingRate, 5600 acoustics, 5601 &inStream); 5602 } 5603 5604 if (inStream != NULL) { 5605 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5606 5607 audio_io_handle_t id = nextUniqueId(); 5608 // Start record thread 5609 // RecorThread require both input and output device indication to forward to audio 5610 // pre processing modules 5611 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5612 thread = new RecordThread(this, 5613 input, 5614 reqSamplingRate, 5615 reqChannels, 5616 id, 5617 device); 5618 mRecordThreads.add(id, thread); 5619 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5620 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5621 if (pFormat != NULL) *pFormat = format; 5622 if (pChannels != NULL) *pChannels = reqChannels; 5623 5624 input->stream->common.standby(&input->stream->common); 5625 5626 // notify client processes of the new input creation 5627 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5628 return id; 5629 } 5630 5631 return 0; 5632} 5633 5634status_t AudioFlinger::closeInput(audio_io_handle_t input) 5635{ 5636 // keep strong reference on the record thread so that 5637 // it is not destroyed while exit() is executed 5638 sp <RecordThread> thread; 5639 { 5640 Mutex::Autolock _l(mLock); 5641 thread = checkRecordThread_l(input); 5642 if (thread == NULL) { 5643 return BAD_VALUE; 5644 } 5645 5646 ALOGV("closeInput() %d", input); 5647 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5648 mRecordThreads.removeItem(input); 5649 } 5650 thread->exit(); 5651 // The thread entity (active unit of execution) is no longer running here, 5652 // but the ThreadBase container still exists. 5653 5654 AudioStreamIn *in = thread->clearInput(); 5655 assert(in != NULL); 5656 // from now on thread->mInput is NULL 5657 in->hwDev->close_input_stream(in->hwDev, in->stream); 5658 delete in; 5659 5660 return NO_ERROR; 5661} 5662 5663status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5664{ 5665 Mutex::Autolock _l(mLock); 5666 MixerThread *dstThread = checkMixerThread_l(output); 5667 if (dstThread == NULL) { 5668 ALOGW("setStreamOutput() bad output id %d", output); 5669 return BAD_VALUE; 5670 } 5671 5672 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5673 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5674 5675 dstThread->setStreamValid(stream, true); 5676 5677 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5678 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5679 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5680 MixerThread *srcThread = (MixerThread *)thread; 5681 srcThread->setStreamValid(stream, false); 5682 srcThread->invalidateTracks(stream); 5683 } 5684 } 5685 5686 return NO_ERROR; 5687} 5688 5689 5690int AudioFlinger::newAudioSessionId() 5691{ 5692 return nextUniqueId(); 5693} 5694 5695void AudioFlinger::acquireAudioSessionId(int audioSession) 5696{ 5697 Mutex::Autolock _l(mLock); 5698 pid_t caller = IPCThreadState::self()->getCallingPid(); 5699 ALOGV("acquiring %d from %d", audioSession, caller); 5700 size_t num = mAudioSessionRefs.size(); 5701 for (size_t i = 0; i< num; i++) { 5702 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5703 if (ref->sessionid == audioSession && ref->pid == caller) { 5704 ref->cnt++; 5705 ALOGV(" incremented refcount to %d", ref->cnt); 5706 return; 5707 } 5708 } 5709 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5710 ALOGV(" added new entry for %d", audioSession); 5711} 5712 5713void AudioFlinger::releaseAudioSessionId(int audioSession) 5714{ 5715 Mutex::Autolock _l(mLock); 5716 pid_t caller = IPCThreadState::self()->getCallingPid(); 5717 ALOGV("releasing %d from %d", audioSession, caller); 5718 size_t num = mAudioSessionRefs.size(); 5719 for (size_t i = 0; i< num; i++) { 5720 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5721 if (ref->sessionid == audioSession && ref->pid == caller) { 5722 ref->cnt--; 5723 ALOGV(" decremented refcount to %d", ref->cnt); 5724 if (ref->cnt == 0) { 5725 mAudioSessionRefs.removeAt(i); 5726 delete ref; 5727 purgeStaleEffects_l(); 5728 } 5729 return; 5730 } 5731 } 5732 ALOGW("session id %d not found for pid %d", audioSession, caller); 5733} 5734 5735void AudioFlinger::purgeStaleEffects_l() { 5736 5737 ALOGV("purging stale effects"); 5738 5739 Vector< sp<EffectChain> > chains; 5740 5741 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5742 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5743 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5744 sp<EffectChain> ec = t->mEffectChains[j]; 5745 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5746 chains.push(ec); 5747 } 5748 } 5749 } 5750 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5751 sp<RecordThread> t = mRecordThreads.valueAt(i); 5752 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5753 sp<EffectChain> ec = t->mEffectChains[j]; 5754 chains.push(ec); 5755 } 5756 } 5757 5758 for (size_t i = 0; i < chains.size(); i++) { 5759 sp<EffectChain> ec = chains[i]; 5760 int sessionid = ec->sessionId(); 5761 sp<ThreadBase> t = ec->mThread.promote(); 5762 if (t == 0) { 5763 continue; 5764 } 5765 size_t numsessionrefs = mAudioSessionRefs.size(); 5766 bool found = false; 5767 for (size_t k = 0; k < numsessionrefs; k++) { 5768 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5769 if (ref->sessionid == sessionid) { 5770 ALOGV(" session %d still exists for %d with %d refs", 5771 sessionid, ref->pid, ref->cnt); 5772 found = true; 5773 break; 5774 } 5775 } 5776 if (!found) { 5777 // remove all effects from the chain 5778 while (ec->mEffects.size()) { 5779 sp<EffectModule> effect = ec->mEffects[0]; 5780 effect->unPin(); 5781 Mutex::Autolock _l (t->mLock); 5782 t->removeEffect_l(effect); 5783 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5784 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5785 if (handle != 0) { 5786 handle->mEffect.clear(); 5787 if (handle->mHasControl && handle->mEnabled) { 5788 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5789 } 5790 } 5791 } 5792 AudioSystem::unregisterEffect(effect->id()); 5793 } 5794 } 5795 } 5796 return; 5797} 5798 5799// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5800AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5801{ 5802 return mPlaybackThreads.valueFor(output).get(); 5803} 5804 5805// checkMixerThread_l() must be called with AudioFlinger::mLock held 5806AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5807{ 5808 PlaybackThread *thread = checkPlaybackThread_l(output); 5809 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5810} 5811 5812// checkRecordThread_l() must be called with AudioFlinger::mLock held 5813AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5814{ 5815 return mRecordThreads.valueFor(input).get(); 5816} 5817 5818uint32_t AudioFlinger::nextUniqueId() 5819{ 5820 return android_atomic_inc(&mNextUniqueId); 5821} 5822 5823AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5824{ 5825 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5826 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5827 AudioStreamOut *output = thread->getOutput(); 5828 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5829 return thread; 5830 } 5831 } 5832 return NULL; 5833} 5834 5835uint32_t AudioFlinger::primaryOutputDevice_l() const 5836{ 5837 PlaybackThread *thread = primaryPlaybackThread_l(); 5838 5839 if (thread == NULL) { 5840 return 0; 5841 } 5842 5843 return thread->device(); 5844} 5845 5846 5847// ---------------------------------------------------------------------------- 5848// Effect management 5849// ---------------------------------------------------------------------------- 5850 5851 5852status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5853{ 5854 Mutex::Autolock _l(mLock); 5855 return EffectQueryNumberEffects(numEffects); 5856} 5857 5858status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5859{ 5860 Mutex::Autolock _l(mLock); 5861 return EffectQueryEffect(index, descriptor); 5862} 5863 5864status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5865 effect_descriptor_t *descriptor) const 5866{ 5867 Mutex::Autolock _l(mLock); 5868 return EffectGetDescriptor(pUuid, descriptor); 5869} 5870 5871 5872sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5873 effect_descriptor_t *pDesc, 5874 const sp<IEffectClient>& effectClient, 5875 int32_t priority, 5876 audio_io_handle_t io, 5877 int sessionId, 5878 status_t *status, 5879 int *id, 5880 int *enabled) 5881{ 5882 status_t lStatus = NO_ERROR; 5883 sp<EffectHandle> handle; 5884 effect_descriptor_t desc; 5885 5886 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5887 pid, effectClient.get(), priority, sessionId, io); 5888 5889 if (pDesc == NULL) { 5890 lStatus = BAD_VALUE; 5891 goto Exit; 5892 } 5893 5894 // check audio settings permission for global effects 5895 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5896 lStatus = PERMISSION_DENIED; 5897 goto Exit; 5898 } 5899 5900 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5901 // that can only be created by audio policy manager (running in same process) 5902 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5903 lStatus = PERMISSION_DENIED; 5904 goto Exit; 5905 } 5906 5907 if (io == 0) { 5908 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5909 // output must be specified by AudioPolicyManager when using session 5910 // AUDIO_SESSION_OUTPUT_STAGE 5911 lStatus = BAD_VALUE; 5912 goto Exit; 5913 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5914 // if the output returned by getOutputForEffect() is removed before we lock the 5915 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5916 // and we will exit safely 5917 io = AudioSystem::getOutputForEffect(&desc); 5918 } 5919 } 5920 5921 { 5922 Mutex::Autolock _l(mLock); 5923 5924 5925 if (!EffectIsNullUuid(&pDesc->uuid)) { 5926 // if uuid is specified, request effect descriptor 5927 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5928 if (lStatus < 0) { 5929 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5930 goto Exit; 5931 } 5932 } else { 5933 // if uuid is not specified, look for an available implementation 5934 // of the required type in effect factory 5935 if (EffectIsNullUuid(&pDesc->type)) { 5936 ALOGW("createEffect() no effect type"); 5937 lStatus = BAD_VALUE; 5938 goto Exit; 5939 } 5940 uint32_t numEffects = 0; 5941 effect_descriptor_t d; 5942 d.flags = 0; // prevent compiler warning 5943 bool found = false; 5944 5945 lStatus = EffectQueryNumberEffects(&numEffects); 5946 if (lStatus < 0) { 5947 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5948 goto Exit; 5949 } 5950 for (uint32_t i = 0; i < numEffects; i++) { 5951 lStatus = EffectQueryEffect(i, &desc); 5952 if (lStatus < 0) { 5953 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5954 continue; 5955 } 5956 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5957 // If matching type found save effect descriptor. If the session is 5958 // 0 and the effect is not auxiliary, continue enumeration in case 5959 // an auxiliary version of this effect type is available 5960 found = true; 5961 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5962 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5963 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5964 break; 5965 } 5966 } 5967 } 5968 if (!found) { 5969 lStatus = BAD_VALUE; 5970 ALOGW("createEffect() effect not found"); 5971 goto Exit; 5972 } 5973 // For same effect type, chose auxiliary version over insert version if 5974 // connect to output mix (Compliance to OpenSL ES) 5975 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5976 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5977 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5978 } 5979 } 5980 5981 // Do not allow auxiliary effects on a session different from 0 (output mix) 5982 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5983 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5984 lStatus = INVALID_OPERATION; 5985 goto Exit; 5986 } 5987 5988 // check recording permission for visualizer 5989 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5990 !recordingAllowed()) { 5991 lStatus = PERMISSION_DENIED; 5992 goto Exit; 5993 } 5994 5995 // return effect descriptor 5996 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5997 5998 // If output is not specified try to find a matching audio session ID in one of the 5999 // output threads. 6000 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6001 // because of code checking output when entering the function. 6002 // Note: io is never 0 when creating an effect on an input 6003 if (io == 0) { 6004 // look for the thread where the specified audio session is present 6005 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6006 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6007 io = mPlaybackThreads.keyAt(i); 6008 break; 6009 } 6010 } 6011 if (io == 0) { 6012 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6013 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6014 io = mRecordThreads.keyAt(i); 6015 break; 6016 } 6017 } 6018 } 6019 // If no output thread contains the requested session ID, default to 6020 // first output. The effect chain will be moved to the correct output 6021 // thread when a track with the same session ID is created 6022 if (io == 0 && mPlaybackThreads.size()) { 6023 io = mPlaybackThreads.keyAt(0); 6024 } 6025 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6026 } 6027 ThreadBase *thread = checkRecordThread_l(io); 6028 if (thread == NULL) { 6029 thread = checkPlaybackThread_l(io); 6030 if (thread == NULL) { 6031 ALOGE("createEffect() unknown output thread"); 6032 lStatus = BAD_VALUE; 6033 goto Exit; 6034 } 6035 } 6036 6037 sp<Client> client = registerPid_l(pid); 6038 6039 // create effect on selected output thread 6040 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6041 &desc, enabled, &lStatus); 6042 if (handle != 0 && id != NULL) { 6043 *id = handle->id(); 6044 } 6045 } 6046 6047Exit: 6048 if(status) { 6049 *status = lStatus; 6050 } 6051 return handle; 6052} 6053 6054status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6055 audio_io_handle_t dstOutput) 6056{ 6057 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6058 sessionId, srcOutput, dstOutput); 6059 Mutex::Autolock _l(mLock); 6060 if (srcOutput == dstOutput) { 6061 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6062 return NO_ERROR; 6063 } 6064 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6065 if (srcThread == NULL) { 6066 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6067 return BAD_VALUE; 6068 } 6069 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6070 if (dstThread == NULL) { 6071 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6072 return BAD_VALUE; 6073 } 6074 6075 Mutex::Autolock _dl(dstThread->mLock); 6076 Mutex::Autolock _sl(srcThread->mLock); 6077 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6078 6079 return NO_ERROR; 6080} 6081 6082// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6083status_t AudioFlinger::moveEffectChain_l(int sessionId, 6084 AudioFlinger::PlaybackThread *srcThread, 6085 AudioFlinger::PlaybackThread *dstThread, 6086 bool reRegister) 6087{ 6088 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6089 sessionId, srcThread, dstThread); 6090 6091 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6092 if (chain == 0) { 6093 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6094 sessionId, srcThread); 6095 return INVALID_OPERATION; 6096 } 6097 6098 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6099 // so that a new chain is created with correct parameters when first effect is added. This is 6100 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6101 // removed. 6102 srcThread->removeEffectChain_l(chain); 6103 6104 // transfer all effects one by one so that new effect chain is created on new thread with 6105 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6106 audio_io_handle_t dstOutput = dstThread->id(); 6107 sp<EffectChain> dstChain; 6108 uint32_t strategy = 0; // prevent compiler warning 6109 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6110 while (effect != 0) { 6111 srcThread->removeEffect_l(effect); 6112 dstThread->addEffect_l(effect); 6113 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6114 if (effect->state() == EffectModule::ACTIVE || 6115 effect->state() == EffectModule::STOPPING) { 6116 effect->start(); 6117 } 6118 // if the move request is not received from audio policy manager, the effect must be 6119 // re-registered with the new strategy and output 6120 if (dstChain == 0) { 6121 dstChain = effect->chain().promote(); 6122 if (dstChain == 0) { 6123 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6124 srcThread->addEffect_l(effect); 6125 return NO_INIT; 6126 } 6127 strategy = dstChain->strategy(); 6128 } 6129 if (reRegister) { 6130 AudioSystem::unregisterEffect(effect->id()); 6131 AudioSystem::registerEffect(&effect->desc(), 6132 dstOutput, 6133 strategy, 6134 sessionId, 6135 effect->id()); 6136 } 6137 effect = chain->getEffectFromId_l(0); 6138 } 6139 6140 return NO_ERROR; 6141} 6142 6143 6144// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6145sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6146 const sp<AudioFlinger::Client>& client, 6147 const sp<IEffectClient>& effectClient, 6148 int32_t priority, 6149 int sessionId, 6150 effect_descriptor_t *desc, 6151 int *enabled, 6152 status_t *status 6153 ) 6154{ 6155 sp<EffectModule> effect; 6156 sp<EffectHandle> handle; 6157 status_t lStatus; 6158 sp<EffectChain> chain; 6159 bool chainCreated = false; 6160 bool effectCreated = false; 6161 bool effectRegistered = false; 6162 6163 lStatus = initCheck(); 6164 if (lStatus != NO_ERROR) { 6165 ALOGW("createEffect_l() Audio driver not initialized."); 6166 goto Exit; 6167 } 6168 6169 // Do not allow effects with session ID 0 on direct output or duplicating threads 6170 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6171 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6172 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6173 desc->name, sessionId); 6174 lStatus = BAD_VALUE; 6175 goto Exit; 6176 } 6177 // Only Pre processor effects are allowed on input threads and only on input threads 6178 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6179 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6180 desc->name, desc->flags, mType); 6181 lStatus = BAD_VALUE; 6182 goto Exit; 6183 } 6184 6185 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6186 6187 { // scope for mLock 6188 Mutex::Autolock _l(mLock); 6189 6190 // check for existing effect chain with the requested audio session 6191 chain = getEffectChain_l(sessionId); 6192 if (chain == 0) { 6193 // create a new chain for this session 6194 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6195 chain = new EffectChain(this, sessionId); 6196 addEffectChain_l(chain); 6197 chain->setStrategy(getStrategyForSession_l(sessionId)); 6198 chainCreated = true; 6199 } else { 6200 effect = chain->getEffectFromDesc_l(desc); 6201 } 6202 6203 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6204 6205 if (effect == 0) { 6206 int id = mAudioFlinger->nextUniqueId(); 6207 // Check CPU and memory usage 6208 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6209 if (lStatus != NO_ERROR) { 6210 goto Exit; 6211 } 6212 effectRegistered = true; 6213 // create a new effect module if none present in the chain 6214 effect = new EffectModule(this, chain, desc, id, sessionId); 6215 lStatus = effect->status(); 6216 if (lStatus != NO_ERROR) { 6217 goto Exit; 6218 } 6219 lStatus = chain->addEffect_l(effect); 6220 if (lStatus != NO_ERROR) { 6221 goto Exit; 6222 } 6223 effectCreated = true; 6224 6225 effect->setDevice(mDevice); 6226 effect->setMode(mAudioFlinger->getMode()); 6227 } 6228 // create effect handle and connect it to effect module 6229 handle = new EffectHandle(effect, client, effectClient, priority); 6230 lStatus = effect->addHandle(handle); 6231 if (enabled != NULL) { 6232 *enabled = (int)effect->isEnabled(); 6233 } 6234 } 6235 6236Exit: 6237 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6238 Mutex::Autolock _l(mLock); 6239 if (effectCreated) { 6240 chain->removeEffect_l(effect); 6241 } 6242 if (effectRegistered) { 6243 AudioSystem::unregisterEffect(effect->id()); 6244 } 6245 if (chainCreated) { 6246 removeEffectChain_l(chain); 6247 } 6248 handle.clear(); 6249 } 6250 6251 if(status) { 6252 *status = lStatus; 6253 } 6254 return handle; 6255} 6256 6257sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6258{ 6259 sp<EffectChain> chain = getEffectChain_l(sessionId); 6260 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6261} 6262 6263// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6264// PlaybackThread::mLock held 6265status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6266{ 6267 // check for existing effect chain with the requested audio session 6268 int sessionId = effect->sessionId(); 6269 sp<EffectChain> chain = getEffectChain_l(sessionId); 6270 bool chainCreated = false; 6271 6272 if (chain == 0) { 6273 // create a new chain for this session 6274 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6275 chain = new EffectChain(this, sessionId); 6276 addEffectChain_l(chain); 6277 chain->setStrategy(getStrategyForSession_l(sessionId)); 6278 chainCreated = true; 6279 } 6280 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6281 6282 if (chain->getEffectFromId_l(effect->id()) != 0) { 6283 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6284 this, effect->desc().name, chain.get()); 6285 return BAD_VALUE; 6286 } 6287 6288 status_t status = chain->addEffect_l(effect); 6289 if (status != NO_ERROR) { 6290 if (chainCreated) { 6291 removeEffectChain_l(chain); 6292 } 6293 return status; 6294 } 6295 6296 effect->setDevice(mDevice); 6297 effect->setMode(mAudioFlinger->getMode()); 6298 return NO_ERROR; 6299} 6300 6301void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6302 6303 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6304 effect_descriptor_t desc = effect->desc(); 6305 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6306 detachAuxEffect_l(effect->id()); 6307 } 6308 6309 sp<EffectChain> chain = effect->chain().promote(); 6310 if (chain != 0) { 6311 // remove effect chain if removing last effect 6312 if (chain->removeEffect_l(effect) == 0) { 6313 removeEffectChain_l(chain); 6314 } 6315 } else { 6316 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6317 } 6318} 6319 6320void AudioFlinger::ThreadBase::lockEffectChains_l( 6321 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6322{ 6323 effectChains = mEffectChains; 6324 for (size_t i = 0; i < mEffectChains.size(); i++) { 6325 mEffectChains[i]->lock(); 6326 } 6327} 6328 6329void AudioFlinger::ThreadBase::unlockEffectChains( 6330 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6331{ 6332 for (size_t i = 0; i < effectChains.size(); i++) { 6333 effectChains[i]->unlock(); 6334 } 6335} 6336 6337sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6338{ 6339 Mutex::Autolock _l(mLock); 6340 return getEffectChain_l(sessionId); 6341} 6342 6343sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6344{ 6345 size_t size = mEffectChains.size(); 6346 for (size_t i = 0; i < size; i++) { 6347 if (mEffectChains[i]->sessionId() == sessionId) { 6348 return mEffectChains[i]; 6349 } 6350 } 6351 return 0; 6352} 6353 6354void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6355{ 6356 Mutex::Autolock _l(mLock); 6357 size_t size = mEffectChains.size(); 6358 for (size_t i = 0; i < size; i++) { 6359 mEffectChains[i]->setMode_l(mode); 6360 } 6361} 6362 6363void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6364 const wp<EffectHandle>& handle, 6365 bool unpinIfLast) { 6366 6367 Mutex::Autolock _l(mLock); 6368 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6369 // delete the effect module if removing last handle on it 6370 if (effect->removeHandle(handle) == 0) { 6371 if (!effect->isPinned() || unpinIfLast) { 6372 removeEffect_l(effect); 6373 AudioSystem::unregisterEffect(effect->id()); 6374 } 6375 } 6376} 6377 6378status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6379{ 6380 int session = chain->sessionId(); 6381 int16_t *buffer = mMixBuffer; 6382 bool ownsBuffer = false; 6383 6384 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6385 if (session > 0) { 6386 // Only one effect chain can be present in direct output thread and it uses 6387 // the mix buffer as input 6388 if (mType != DIRECT) { 6389 size_t numSamples = mFrameCount * mChannelCount; 6390 buffer = new int16_t[numSamples]; 6391 memset(buffer, 0, numSamples * sizeof(int16_t)); 6392 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6393 ownsBuffer = true; 6394 } 6395 6396 // Attach all tracks with same session ID to this chain. 6397 for (size_t i = 0; i < mTracks.size(); ++i) { 6398 sp<Track> track = mTracks[i]; 6399 if (session == track->sessionId()) { 6400 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6401 track->setMainBuffer(buffer); 6402 chain->incTrackCnt(); 6403 } 6404 } 6405 6406 // indicate all active tracks in the chain 6407 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6408 sp<Track> track = mActiveTracks[i].promote(); 6409 if (track == 0) continue; 6410 if (session == track->sessionId()) { 6411 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6412 chain->incActiveTrackCnt(); 6413 } 6414 } 6415 } 6416 6417 chain->setInBuffer(buffer, ownsBuffer); 6418 chain->setOutBuffer(mMixBuffer); 6419 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6420 // chains list in order to be processed last as it contains output stage effects 6421 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6422 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6423 // after track specific effects and before output stage 6424 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6425 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6426 // Effect chain for other sessions are inserted at beginning of effect 6427 // chains list to be processed before output mix effects. Relative order between other 6428 // sessions is not important 6429 size_t size = mEffectChains.size(); 6430 size_t i = 0; 6431 for (i = 0; i < size; i++) { 6432 if (mEffectChains[i]->sessionId() < session) break; 6433 } 6434 mEffectChains.insertAt(chain, i); 6435 checkSuspendOnAddEffectChain_l(chain); 6436 6437 return NO_ERROR; 6438} 6439 6440size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6441{ 6442 int session = chain->sessionId(); 6443 6444 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6445 6446 for (size_t i = 0; i < mEffectChains.size(); i++) { 6447 if (chain == mEffectChains[i]) { 6448 mEffectChains.removeAt(i); 6449 // detach all active tracks from the chain 6450 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6451 sp<Track> track = mActiveTracks[i].promote(); 6452 if (track == 0) continue; 6453 if (session == track->sessionId()) { 6454 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6455 chain.get(), session); 6456 chain->decActiveTrackCnt(); 6457 } 6458 } 6459 6460 // detach all tracks with same session ID from this chain 6461 for (size_t i = 0; i < mTracks.size(); ++i) { 6462 sp<Track> track = mTracks[i]; 6463 if (session == track->sessionId()) { 6464 track->setMainBuffer(mMixBuffer); 6465 chain->decTrackCnt(); 6466 } 6467 } 6468 break; 6469 } 6470 } 6471 return mEffectChains.size(); 6472} 6473 6474status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6475 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6476{ 6477 Mutex::Autolock _l(mLock); 6478 return attachAuxEffect_l(track, EffectId); 6479} 6480 6481status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6482 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6483{ 6484 status_t status = NO_ERROR; 6485 6486 if (EffectId == 0) { 6487 track->setAuxBuffer(0, NULL); 6488 } else { 6489 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6490 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6491 if (effect != 0) { 6492 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6493 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6494 } else { 6495 status = INVALID_OPERATION; 6496 } 6497 } else { 6498 status = BAD_VALUE; 6499 } 6500 } 6501 return status; 6502} 6503 6504void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6505{ 6506 for (size_t i = 0; i < mTracks.size(); ++i) { 6507 sp<Track> track = mTracks[i]; 6508 if (track->auxEffectId() == effectId) { 6509 attachAuxEffect_l(track, 0); 6510 } 6511 } 6512} 6513 6514status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6515{ 6516 // only one chain per input thread 6517 if (mEffectChains.size() != 0) { 6518 return INVALID_OPERATION; 6519 } 6520 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6521 6522 chain->setInBuffer(NULL); 6523 chain->setOutBuffer(NULL); 6524 6525 checkSuspendOnAddEffectChain_l(chain); 6526 6527 mEffectChains.add(chain); 6528 6529 return NO_ERROR; 6530} 6531 6532size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6533{ 6534 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6535 ALOGW_IF(mEffectChains.size() != 1, 6536 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6537 chain.get(), mEffectChains.size(), this); 6538 if (mEffectChains.size() == 1) { 6539 mEffectChains.removeAt(0); 6540 } 6541 return 0; 6542} 6543 6544// ---------------------------------------------------------------------------- 6545// EffectModule implementation 6546// ---------------------------------------------------------------------------- 6547 6548#undef LOG_TAG 6549#define LOG_TAG "AudioFlinger::EffectModule" 6550 6551AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6552 const wp<AudioFlinger::EffectChain>& chain, 6553 effect_descriptor_t *desc, 6554 int id, 6555 int sessionId) 6556 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6557 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6558{ 6559 ALOGV("Constructor %p", this); 6560 int lStatus; 6561 if (thread == NULL) { 6562 return; 6563 } 6564 6565 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6566 6567 // create effect engine from effect factory 6568 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6569 6570 if (mStatus != NO_ERROR) { 6571 return; 6572 } 6573 lStatus = init(); 6574 if (lStatus < 0) { 6575 mStatus = lStatus; 6576 goto Error; 6577 } 6578 6579 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6580 mPinned = true; 6581 } 6582 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6583 return; 6584Error: 6585 EffectRelease(mEffectInterface); 6586 mEffectInterface = NULL; 6587 ALOGV("Constructor Error %d", mStatus); 6588} 6589 6590AudioFlinger::EffectModule::~EffectModule() 6591{ 6592 ALOGV("Destructor %p", this); 6593 if (mEffectInterface != NULL) { 6594 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6595 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6596 sp<ThreadBase> thread = mThread.promote(); 6597 if (thread != 0) { 6598 audio_stream_t *stream = thread->stream(); 6599 if (stream != NULL) { 6600 stream->remove_audio_effect(stream, mEffectInterface); 6601 } 6602 } 6603 } 6604 // release effect engine 6605 EffectRelease(mEffectInterface); 6606 } 6607} 6608 6609status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6610{ 6611 status_t status; 6612 6613 Mutex::Autolock _l(mLock); 6614 int priority = handle->priority(); 6615 size_t size = mHandles.size(); 6616 sp<EffectHandle> h; 6617 size_t i; 6618 for (i = 0; i < size; i++) { 6619 h = mHandles[i].promote(); 6620 if (h == 0) continue; 6621 if (h->priority() <= priority) break; 6622 } 6623 // if inserted in first place, move effect control from previous owner to this handle 6624 if (i == 0) { 6625 bool enabled = false; 6626 if (h != 0) { 6627 enabled = h->enabled(); 6628 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6629 } 6630 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6631 status = NO_ERROR; 6632 } else { 6633 status = ALREADY_EXISTS; 6634 } 6635 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6636 mHandles.insertAt(handle, i); 6637 return status; 6638} 6639 6640size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6641{ 6642 Mutex::Autolock _l(mLock); 6643 size_t size = mHandles.size(); 6644 size_t i; 6645 for (i = 0; i < size; i++) { 6646 if (mHandles[i] == handle) break; 6647 } 6648 if (i == size) { 6649 return size; 6650 } 6651 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6652 6653 bool enabled = false; 6654 EffectHandle *hdl = handle.unsafe_get(); 6655 if (hdl != NULL) { 6656 ALOGV("removeHandle() unsafe_get OK"); 6657 enabled = hdl->enabled(); 6658 } 6659 mHandles.removeAt(i); 6660 size = mHandles.size(); 6661 // if removed from first place, move effect control from this handle to next in line 6662 if (i == 0 && size != 0) { 6663 sp<EffectHandle> h = mHandles[0].promote(); 6664 if (h != 0) { 6665 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6666 } 6667 } 6668 6669 // Prevent calls to process() and other functions on effect interface from now on. 6670 // The effect engine will be released by the destructor when the last strong reference on 6671 // this object is released which can happen after next process is called. 6672 if (size == 0 && !mPinned) { 6673 mState = DESTROYED; 6674 } 6675 6676 return size; 6677} 6678 6679sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6680{ 6681 Mutex::Autolock _l(mLock); 6682 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6683} 6684 6685void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6686{ 6687 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6688 // keep a strong reference on this EffectModule to avoid calling the 6689 // destructor before we exit 6690 sp<EffectModule> keep(this); 6691 { 6692 sp<ThreadBase> thread = mThread.promote(); 6693 if (thread != 0) { 6694 thread->disconnectEffect(keep, handle, unpinIfLast); 6695 } 6696 } 6697} 6698 6699void AudioFlinger::EffectModule::updateState() { 6700 Mutex::Autolock _l(mLock); 6701 6702 switch (mState) { 6703 case RESTART: 6704 reset_l(); 6705 // FALL THROUGH 6706 6707 case STARTING: 6708 // clear auxiliary effect input buffer for next accumulation 6709 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6710 memset(mConfig.inputCfg.buffer.raw, 6711 0, 6712 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6713 } 6714 start_l(); 6715 mState = ACTIVE; 6716 break; 6717 case STOPPING: 6718 stop_l(); 6719 mDisableWaitCnt = mMaxDisableWaitCnt; 6720 mState = STOPPED; 6721 break; 6722 case STOPPED: 6723 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6724 // turn off sequence. 6725 if (--mDisableWaitCnt == 0) { 6726 reset_l(); 6727 mState = IDLE; 6728 } 6729 break; 6730 default: //IDLE , ACTIVE, DESTROYED 6731 break; 6732 } 6733} 6734 6735void AudioFlinger::EffectModule::process() 6736{ 6737 Mutex::Autolock _l(mLock); 6738 6739 if (mState == DESTROYED || mEffectInterface == NULL || 6740 mConfig.inputCfg.buffer.raw == NULL || 6741 mConfig.outputCfg.buffer.raw == NULL) { 6742 return; 6743 } 6744 6745 if (isProcessEnabled()) { 6746 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6747 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6748 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6749 mConfig.inputCfg.buffer.s32, 6750 mConfig.inputCfg.buffer.frameCount/2); 6751 } 6752 6753 // do the actual processing in the effect engine 6754 int ret = (*mEffectInterface)->process(mEffectInterface, 6755 &mConfig.inputCfg.buffer, 6756 &mConfig.outputCfg.buffer); 6757 6758 // force transition to IDLE state when engine is ready 6759 if (mState == STOPPED && ret == -ENODATA) { 6760 mDisableWaitCnt = 1; 6761 } 6762 6763 // clear auxiliary effect input buffer for next accumulation 6764 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6765 memset(mConfig.inputCfg.buffer.raw, 0, 6766 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6767 } 6768 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6769 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6770 // If an insert effect is idle and input buffer is different from output buffer, 6771 // accumulate input onto output 6772 sp<EffectChain> chain = mChain.promote(); 6773 if (chain != 0 && chain->activeTrackCnt() != 0) { 6774 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6775 int16_t *in = mConfig.inputCfg.buffer.s16; 6776 int16_t *out = mConfig.outputCfg.buffer.s16; 6777 for (size_t i = 0; i < frameCnt; i++) { 6778 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6779 } 6780 } 6781 } 6782} 6783 6784void AudioFlinger::EffectModule::reset_l() 6785{ 6786 if (mEffectInterface == NULL) { 6787 return; 6788 } 6789 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6790} 6791 6792status_t AudioFlinger::EffectModule::configure() 6793{ 6794 uint32_t channels; 6795 if (mEffectInterface == NULL) { 6796 return NO_INIT; 6797 } 6798 6799 sp<ThreadBase> thread = mThread.promote(); 6800 if (thread == 0) { 6801 return DEAD_OBJECT; 6802 } 6803 6804 // TODO: handle configuration of effects replacing track process 6805 if (thread->channelCount() == 1) { 6806 channels = AUDIO_CHANNEL_OUT_MONO; 6807 } else { 6808 channels = AUDIO_CHANNEL_OUT_STEREO; 6809 } 6810 6811 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6812 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6813 } else { 6814 mConfig.inputCfg.channels = channels; 6815 } 6816 mConfig.outputCfg.channels = channels; 6817 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6818 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6819 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6820 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6821 mConfig.inputCfg.bufferProvider.cookie = NULL; 6822 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6823 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6824 mConfig.outputCfg.bufferProvider.cookie = NULL; 6825 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6826 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6827 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6828 // Insert effect: 6829 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6830 // always overwrites output buffer: input buffer == output buffer 6831 // - in other sessions: 6832 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6833 // other effect: overwrites output buffer: input buffer == output buffer 6834 // Auxiliary effect: 6835 // accumulates in output buffer: input buffer != output buffer 6836 // Therefore: accumulate <=> input buffer != output buffer 6837 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6838 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6839 } else { 6840 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6841 } 6842 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6843 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6844 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6845 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6846 6847 ALOGV("configure() %p thread %p buffer %p framecount %d", 6848 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6849 6850 status_t cmdStatus; 6851 uint32_t size = sizeof(int); 6852 status_t status = (*mEffectInterface)->command(mEffectInterface, 6853 EFFECT_CMD_SET_CONFIG, 6854 sizeof(effect_config_t), 6855 &mConfig, 6856 &size, 6857 &cmdStatus); 6858 if (status == 0) { 6859 status = cmdStatus; 6860 } 6861 6862 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6863 (1000 * mConfig.outputCfg.buffer.frameCount); 6864 6865 return status; 6866} 6867 6868status_t AudioFlinger::EffectModule::init() 6869{ 6870 Mutex::Autolock _l(mLock); 6871 if (mEffectInterface == NULL) { 6872 return NO_INIT; 6873 } 6874 status_t cmdStatus; 6875 uint32_t size = sizeof(status_t); 6876 status_t status = (*mEffectInterface)->command(mEffectInterface, 6877 EFFECT_CMD_INIT, 6878 0, 6879 NULL, 6880 &size, 6881 &cmdStatus); 6882 if (status == 0) { 6883 status = cmdStatus; 6884 } 6885 return status; 6886} 6887 6888status_t AudioFlinger::EffectModule::start() 6889{ 6890 Mutex::Autolock _l(mLock); 6891 return start_l(); 6892} 6893 6894status_t AudioFlinger::EffectModule::start_l() 6895{ 6896 if (mEffectInterface == NULL) { 6897 return NO_INIT; 6898 } 6899 status_t cmdStatus; 6900 uint32_t size = sizeof(status_t); 6901 status_t status = (*mEffectInterface)->command(mEffectInterface, 6902 EFFECT_CMD_ENABLE, 6903 0, 6904 NULL, 6905 &size, 6906 &cmdStatus); 6907 if (status == 0) { 6908 status = cmdStatus; 6909 } 6910 if (status == 0 && 6911 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6912 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6913 sp<ThreadBase> thread = mThread.promote(); 6914 if (thread != 0) { 6915 audio_stream_t *stream = thread->stream(); 6916 if (stream != NULL) { 6917 stream->add_audio_effect(stream, mEffectInterface); 6918 } 6919 } 6920 } 6921 return status; 6922} 6923 6924status_t AudioFlinger::EffectModule::stop() 6925{ 6926 Mutex::Autolock _l(mLock); 6927 return stop_l(); 6928} 6929 6930status_t AudioFlinger::EffectModule::stop_l() 6931{ 6932 if (mEffectInterface == NULL) { 6933 return NO_INIT; 6934 } 6935 status_t cmdStatus; 6936 uint32_t size = sizeof(status_t); 6937 status_t status = (*mEffectInterface)->command(mEffectInterface, 6938 EFFECT_CMD_DISABLE, 6939 0, 6940 NULL, 6941 &size, 6942 &cmdStatus); 6943 if (status == 0) { 6944 status = cmdStatus; 6945 } 6946 if (status == 0 && 6947 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6948 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6949 sp<ThreadBase> thread = mThread.promote(); 6950 if (thread != 0) { 6951 audio_stream_t *stream = thread->stream(); 6952 if (stream != NULL) { 6953 stream->remove_audio_effect(stream, mEffectInterface); 6954 } 6955 } 6956 } 6957 return status; 6958} 6959 6960status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6961 uint32_t cmdSize, 6962 void *pCmdData, 6963 uint32_t *replySize, 6964 void *pReplyData) 6965{ 6966 Mutex::Autolock _l(mLock); 6967// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6968 6969 if (mState == DESTROYED || mEffectInterface == NULL) { 6970 return NO_INIT; 6971 } 6972 status_t status = (*mEffectInterface)->command(mEffectInterface, 6973 cmdCode, 6974 cmdSize, 6975 pCmdData, 6976 replySize, 6977 pReplyData); 6978 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6979 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6980 for (size_t i = 1; i < mHandles.size(); i++) { 6981 sp<EffectHandle> h = mHandles[i].promote(); 6982 if (h != 0) { 6983 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6984 } 6985 } 6986 } 6987 return status; 6988} 6989 6990status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6991{ 6992 6993 Mutex::Autolock _l(mLock); 6994 ALOGV("setEnabled %p enabled %d", this, enabled); 6995 6996 if (enabled != isEnabled()) { 6997 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6998 if (enabled && status != NO_ERROR) { 6999 return status; 7000 } 7001 7002 switch (mState) { 7003 // going from disabled to enabled 7004 case IDLE: 7005 mState = STARTING; 7006 break; 7007 case STOPPED: 7008 mState = RESTART; 7009 break; 7010 case STOPPING: 7011 mState = ACTIVE; 7012 break; 7013 7014 // going from enabled to disabled 7015 case RESTART: 7016 mState = STOPPED; 7017 break; 7018 case STARTING: 7019 mState = IDLE; 7020 break; 7021 case ACTIVE: 7022 mState = STOPPING; 7023 break; 7024 case DESTROYED: 7025 return NO_ERROR; // simply ignore as we are being destroyed 7026 } 7027 for (size_t i = 1; i < mHandles.size(); i++) { 7028 sp<EffectHandle> h = mHandles[i].promote(); 7029 if (h != 0) { 7030 h->setEnabled(enabled); 7031 } 7032 } 7033 } 7034 return NO_ERROR; 7035} 7036 7037bool AudioFlinger::EffectModule::isEnabled() const 7038{ 7039 switch (mState) { 7040 case RESTART: 7041 case STARTING: 7042 case ACTIVE: 7043 return true; 7044 case IDLE: 7045 case STOPPING: 7046 case STOPPED: 7047 case DESTROYED: 7048 default: 7049 return false; 7050 } 7051} 7052 7053bool AudioFlinger::EffectModule::isProcessEnabled() const 7054{ 7055 switch (mState) { 7056 case RESTART: 7057 case ACTIVE: 7058 case STOPPING: 7059 case STOPPED: 7060 return true; 7061 case IDLE: 7062 case STARTING: 7063 case DESTROYED: 7064 default: 7065 return false; 7066 } 7067} 7068 7069status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7070{ 7071 Mutex::Autolock _l(mLock); 7072 status_t status = NO_ERROR; 7073 7074 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7075 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7076 if (isProcessEnabled() && 7077 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7078 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7079 status_t cmdStatus; 7080 uint32_t volume[2]; 7081 uint32_t *pVolume = NULL; 7082 uint32_t size = sizeof(volume); 7083 volume[0] = *left; 7084 volume[1] = *right; 7085 if (controller) { 7086 pVolume = volume; 7087 } 7088 status = (*mEffectInterface)->command(mEffectInterface, 7089 EFFECT_CMD_SET_VOLUME, 7090 size, 7091 volume, 7092 &size, 7093 pVolume); 7094 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7095 *left = volume[0]; 7096 *right = volume[1]; 7097 } 7098 } 7099 return status; 7100} 7101 7102status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7103{ 7104 Mutex::Autolock _l(mLock); 7105 status_t status = NO_ERROR; 7106 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7107 // audio pre processing modules on RecordThread can receive both output and 7108 // input device indication in the same call 7109 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7110 if (dev) { 7111 status_t cmdStatus; 7112 uint32_t size = sizeof(status_t); 7113 7114 status = (*mEffectInterface)->command(mEffectInterface, 7115 EFFECT_CMD_SET_DEVICE, 7116 sizeof(uint32_t), 7117 &dev, 7118 &size, 7119 &cmdStatus); 7120 if (status == NO_ERROR) { 7121 status = cmdStatus; 7122 } 7123 } 7124 dev = device & AUDIO_DEVICE_IN_ALL; 7125 if (dev) { 7126 status_t cmdStatus; 7127 uint32_t size = sizeof(status_t); 7128 7129 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7130 EFFECT_CMD_SET_INPUT_DEVICE, 7131 sizeof(uint32_t), 7132 &dev, 7133 &size, 7134 &cmdStatus); 7135 if (status2 == NO_ERROR) { 7136 status2 = cmdStatus; 7137 } 7138 if (status == NO_ERROR) { 7139 status = status2; 7140 } 7141 } 7142 } 7143 return status; 7144} 7145 7146status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7147{ 7148 Mutex::Autolock _l(mLock); 7149 status_t status = NO_ERROR; 7150 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7151 status_t cmdStatus; 7152 uint32_t size = sizeof(status_t); 7153 status = (*mEffectInterface)->command(mEffectInterface, 7154 EFFECT_CMD_SET_AUDIO_MODE, 7155 sizeof(audio_mode_t), 7156 &mode, 7157 &size, 7158 &cmdStatus); 7159 if (status == NO_ERROR) { 7160 status = cmdStatus; 7161 } 7162 } 7163 return status; 7164} 7165 7166void AudioFlinger::EffectModule::setSuspended(bool suspended) 7167{ 7168 Mutex::Autolock _l(mLock); 7169 mSuspended = suspended; 7170} 7171 7172bool AudioFlinger::EffectModule::suspended() const 7173{ 7174 Mutex::Autolock _l(mLock); 7175 return mSuspended; 7176} 7177 7178status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7179{ 7180 const size_t SIZE = 256; 7181 char buffer[SIZE]; 7182 String8 result; 7183 7184 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7185 result.append(buffer); 7186 7187 bool locked = tryLock(mLock); 7188 // failed to lock - AudioFlinger is probably deadlocked 7189 if (!locked) { 7190 result.append("\t\tCould not lock Fx mutex:\n"); 7191 } 7192 7193 result.append("\t\tSession Status State Engine:\n"); 7194 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7195 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7196 result.append(buffer); 7197 7198 result.append("\t\tDescriptor:\n"); 7199 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7200 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7201 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7202 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7203 result.append(buffer); 7204 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7205 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7206 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7207 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7208 result.append(buffer); 7209 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7210 mDescriptor.apiVersion, 7211 mDescriptor.flags); 7212 result.append(buffer); 7213 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7214 mDescriptor.name); 7215 result.append(buffer); 7216 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7217 mDescriptor.implementor); 7218 result.append(buffer); 7219 7220 result.append("\t\t- Input configuration:\n"); 7221 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7222 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7223 (uint32_t)mConfig.inputCfg.buffer.raw, 7224 mConfig.inputCfg.buffer.frameCount, 7225 mConfig.inputCfg.samplingRate, 7226 mConfig.inputCfg.channels, 7227 mConfig.inputCfg.format); 7228 result.append(buffer); 7229 7230 result.append("\t\t- Output configuration:\n"); 7231 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7232 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7233 (uint32_t)mConfig.outputCfg.buffer.raw, 7234 mConfig.outputCfg.buffer.frameCount, 7235 mConfig.outputCfg.samplingRate, 7236 mConfig.outputCfg.channels, 7237 mConfig.outputCfg.format); 7238 result.append(buffer); 7239 7240 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7241 result.append(buffer); 7242 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7243 for (size_t i = 0; i < mHandles.size(); ++i) { 7244 sp<EffectHandle> handle = mHandles[i].promote(); 7245 if (handle != 0) { 7246 handle->dump(buffer, SIZE); 7247 result.append(buffer); 7248 } 7249 } 7250 7251 result.append("\n"); 7252 7253 write(fd, result.string(), result.length()); 7254 7255 if (locked) { 7256 mLock.unlock(); 7257 } 7258 7259 return NO_ERROR; 7260} 7261 7262// ---------------------------------------------------------------------------- 7263// EffectHandle implementation 7264// ---------------------------------------------------------------------------- 7265 7266#undef LOG_TAG 7267#define LOG_TAG "AudioFlinger::EffectHandle" 7268 7269AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7270 const sp<AudioFlinger::Client>& client, 7271 const sp<IEffectClient>& effectClient, 7272 int32_t priority) 7273 : BnEffect(), 7274 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7275 mPriority(priority), mHasControl(false), mEnabled(false) 7276{ 7277 ALOGV("constructor %p", this); 7278 7279 if (client == 0) { 7280 return; 7281 } 7282 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7283 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7284 if (mCblkMemory != 0) { 7285 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7286 7287 if (mCblk != NULL) { 7288 new(mCblk) effect_param_cblk_t(); 7289 mBuffer = (uint8_t *)mCblk + bufOffset; 7290 } 7291 } else { 7292 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7293 return; 7294 } 7295} 7296 7297AudioFlinger::EffectHandle::~EffectHandle() 7298{ 7299 ALOGV("Destructor %p", this); 7300 disconnect(false); 7301 ALOGV("Destructor DONE %p", this); 7302} 7303 7304status_t AudioFlinger::EffectHandle::enable() 7305{ 7306 ALOGV("enable %p", this); 7307 if (!mHasControl) return INVALID_OPERATION; 7308 if (mEffect == 0) return DEAD_OBJECT; 7309 7310 if (mEnabled) { 7311 return NO_ERROR; 7312 } 7313 7314 mEnabled = true; 7315 7316 sp<ThreadBase> thread = mEffect->thread().promote(); 7317 if (thread != 0) { 7318 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7319 } 7320 7321 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7322 if (mEffect->suspended()) { 7323 return NO_ERROR; 7324 } 7325 7326 status_t status = mEffect->setEnabled(true); 7327 if (status != NO_ERROR) { 7328 if (thread != 0) { 7329 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7330 } 7331 mEnabled = false; 7332 } 7333 return status; 7334} 7335 7336status_t AudioFlinger::EffectHandle::disable() 7337{ 7338 ALOGV("disable %p", this); 7339 if (!mHasControl) return INVALID_OPERATION; 7340 if (mEffect == 0) return DEAD_OBJECT; 7341 7342 if (!mEnabled) { 7343 return NO_ERROR; 7344 } 7345 mEnabled = false; 7346 7347 if (mEffect->suspended()) { 7348 return NO_ERROR; 7349 } 7350 7351 status_t status = mEffect->setEnabled(false); 7352 7353 sp<ThreadBase> thread = mEffect->thread().promote(); 7354 if (thread != 0) { 7355 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7356 } 7357 7358 return status; 7359} 7360 7361void AudioFlinger::EffectHandle::disconnect() 7362{ 7363 disconnect(true); 7364} 7365 7366void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7367{ 7368 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7369 if (mEffect == 0) { 7370 return; 7371 } 7372 mEffect->disconnect(this, unpinIfLast); 7373 7374 if (mHasControl && mEnabled) { 7375 sp<ThreadBase> thread = mEffect->thread().promote(); 7376 if (thread != 0) { 7377 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7378 } 7379 } 7380 7381 // release sp on module => module destructor can be called now 7382 mEffect.clear(); 7383 if (mClient != 0) { 7384 if (mCblk != NULL) { 7385 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7386 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7387 } 7388 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7389 // Client destructor must run with AudioFlinger mutex locked 7390 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7391 mClient.clear(); 7392 } 7393} 7394 7395status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7396 uint32_t cmdSize, 7397 void *pCmdData, 7398 uint32_t *replySize, 7399 void *pReplyData) 7400{ 7401// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7402// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7403 7404 // only get parameter command is permitted for applications not controlling the effect 7405 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7406 return INVALID_OPERATION; 7407 } 7408 if (mEffect == 0) return DEAD_OBJECT; 7409 if (mClient == 0) return INVALID_OPERATION; 7410 7411 // handle commands that are not forwarded transparently to effect engine 7412 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7413 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7414 // no risk to block the whole media server process or mixer threads is we are stuck here 7415 Mutex::Autolock _l(mCblk->lock); 7416 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7417 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7418 mCblk->serverIndex = 0; 7419 mCblk->clientIndex = 0; 7420 return BAD_VALUE; 7421 } 7422 status_t status = NO_ERROR; 7423 while (mCblk->serverIndex < mCblk->clientIndex) { 7424 int reply; 7425 uint32_t rsize = sizeof(int); 7426 int *p = (int *)(mBuffer + mCblk->serverIndex); 7427 int size = *p++; 7428 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7429 ALOGW("command(): invalid parameter block size"); 7430 break; 7431 } 7432 effect_param_t *param = (effect_param_t *)p; 7433 if (param->psize == 0 || param->vsize == 0) { 7434 ALOGW("command(): null parameter or value size"); 7435 mCblk->serverIndex += size; 7436 continue; 7437 } 7438 uint32_t psize = sizeof(effect_param_t) + 7439 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7440 param->vsize; 7441 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7442 psize, 7443 p, 7444 &rsize, 7445 &reply); 7446 // stop at first error encountered 7447 if (ret != NO_ERROR) { 7448 status = ret; 7449 *(int *)pReplyData = reply; 7450 break; 7451 } else if (reply != NO_ERROR) { 7452 *(int *)pReplyData = reply; 7453 break; 7454 } 7455 mCblk->serverIndex += size; 7456 } 7457 mCblk->serverIndex = 0; 7458 mCblk->clientIndex = 0; 7459 return status; 7460 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7461 *(int *)pReplyData = NO_ERROR; 7462 return enable(); 7463 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7464 *(int *)pReplyData = NO_ERROR; 7465 return disable(); 7466 } 7467 7468 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7469} 7470 7471void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7472{ 7473 ALOGV("setControl %p control %d", this, hasControl); 7474 7475 mHasControl = hasControl; 7476 mEnabled = enabled; 7477 7478 if (signal && mEffectClient != 0) { 7479 mEffectClient->controlStatusChanged(hasControl); 7480 } 7481} 7482 7483void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7484 uint32_t cmdSize, 7485 void *pCmdData, 7486 uint32_t replySize, 7487 void *pReplyData) 7488{ 7489 if (mEffectClient != 0) { 7490 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7491 } 7492} 7493 7494 7495 7496void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7497{ 7498 if (mEffectClient != 0) { 7499 mEffectClient->enableStatusChanged(enabled); 7500 } 7501} 7502 7503status_t AudioFlinger::EffectHandle::onTransact( 7504 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7505{ 7506 return BnEffect::onTransact(code, data, reply, flags); 7507} 7508 7509 7510void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7511{ 7512 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7513 7514 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7515 (mClient == 0) ? getpid_cached : mClient->pid(), 7516 mPriority, 7517 mHasControl, 7518 !locked, 7519 mCblk ? mCblk->clientIndex : 0, 7520 mCblk ? mCblk->serverIndex : 0 7521 ); 7522 7523 if (locked) { 7524 mCblk->lock.unlock(); 7525 } 7526} 7527 7528#undef LOG_TAG 7529#define LOG_TAG "AudioFlinger::EffectChain" 7530 7531AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7532 int sessionId) 7533 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7534 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7535 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7536{ 7537 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7538 if (thread == NULL) { 7539 return; 7540 } 7541 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7542 thread->frameCount(); 7543} 7544 7545AudioFlinger::EffectChain::~EffectChain() 7546{ 7547 if (mOwnInBuffer) { 7548 delete mInBuffer; 7549 } 7550 7551} 7552 7553// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7554sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7555{ 7556 size_t size = mEffects.size(); 7557 7558 for (size_t i = 0; i < size; i++) { 7559 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7560 return mEffects[i]; 7561 } 7562 } 7563 return 0; 7564} 7565 7566// getEffectFromId_l() must be called with ThreadBase::mLock held 7567sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7568{ 7569 size_t size = mEffects.size(); 7570 7571 for (size_t i = 0; i < size; i++) { 7572 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7573 if (id == 0 || mEffects[i]->id() == id) { 7574 return mEffects[i]; 7575 } 7576 } 7577 return 0; 7578} 7579 7580// getEffectFromType_l() must be called with ThreadBase::mLock held 7581sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7582 const effect_uuid_t *type) 7583{ 7584 size_t size = mEffects.size(); 7585 7586 for (size_t i = 0; i < size; i++) { 7587 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7588 return mEffects[i]; 7589 } 7590 } 7591 return 0; 7592} 7593 7594// Must be called with EffectChain::mLock locked 7595void AudioFlinger::EffectChain::process_l() 7596{ 7597 sp<ThreadBase> thread = mThread.promote(); 7598 if (thread == 0) { 7599 ALOGW("process_l(): cannot promote mixer thread"); 7600 return; 7601 } 7602 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7603 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7604 // always process effects unless no more tracks are on the session and the effect tail 7605 // has been rendered 7606 bool doProcess = true; 7607 if (!isGlobalSession) { 7608 bool tracksOnSession = (trackCnt() != 0); 7609 7610 if (!tracksOnSession && mTailBufferCount == 0) { 7611 doProcess = false; 7612 } 7613 7614 if (activeTrackCnt() == 0) { 7615 // if no track is active and the effect tail has not been rendered, 7616 // the input buffer must be cleared here as the mixer process will not do it 7617 if (tracksOnSession || mTailBufferCount > 0) { 7618 size_t numSamples = thread->frameCount() * thread->channelCount(); 7619 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7620 if (mTailBufferCount > 0) { 7621 mTailBufferCount--; 7622 } 7623 } 7624 } 7625 } 7626 7627 size_t size = mEffects.size(); 7628 if (doProcess) { 7629 for (size_t i = 0; i < size; i++) { 7630 mEffects[i]->process(); 7631 } 7632 } 7633 for (size_t i = 0; i < size; i++) { 7634 mEffects[i]->updateState(); 7635 } 7636} 7637 7638// addEffect_l() must be called with PlaybackThread::mLock held 7639status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7640{ 7641 effect_descriptor_t desc = effect->desc(); 7642 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7643 7644 Mutex::Autolock _l(mLock); 7645 effect->setChain(this); 7646 sp<ThreadBase> thread = mThread.promote(); 7647 if (thread == 0) { 7648 return NO_INIT; 7649 } 7650 effect->setThread(thread); 7651 7652 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7653 // Auxiliary effects are inserted at the beginning of mEffects vector as 7654 // they are processed first and accumulated in chain input buffer 7655 mEffects.insertAt(effect, 0); 7656 7657 // the input buffer for auxiliary effect contains mono samples in 7658 // 32 bit format. This is to avoid saturation in AudoMixer 7659 // accumulation stage. Saturation is done in EffectModule::process() before 7660 // calling the process in effect engine 7661 size_t numSamples = thread->frameCount(); 7662 int32_t *buffer = new int32_t[numSamples]; 7663 memset(buffer, 0, numSamples * sizeof(int32_t)); 7664 effect->setInBuffer((int16_t *)buffer); 7665 // auxiliary effects output samples to chain input buffer for further processing 7666 // by insert effects 7667 effect->setOutBuffer(mInBuffer); 7668 } else { 7669 // Insert effects are inserted at the end of mEffects vector as they are processed 7670 // after track and auxiliary effects. 7671 // Insert effect order as a function of indicated preference: 7672 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7673 // another effect is present 7674 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7675 // last effect claiming first position 7676 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7677 // first effect claiming last position 7678 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7679 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7680 // already present 7681 7682 size_t size = mEffects.size(); 7683 size_t idx_insert = size; 7684 ssize_t idx_insert_first = -1; 7685 ssize_t idx_insert_last = -1; 7686 7687 for (size_t i = 0; i < size; i++) { 7688 effect_descriptor_t d = mEffects[i]->desc(); 7689 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7690 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7691 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7692 // check invalid effect chaining combinations 7693 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7694 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7695 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7696 return INVALID_OPERATION; 7697 } 7698 // remember position of first insert effect and by default 7699 // select this as insert position for new effect 7700 if (idx_insert == size) { 7701 idx_insert = i; 7702 } 7703 // remember position of last insert effect claiming 7704 // first position 7705 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7706 idx_insert_first = i; 7707 } 7708 // remember position of first insert effect claiming 7709 // last position 7710 if (iPref == EFFECT_FLAG_INSERT_LAST && 7711 idx_insert_last == -1) { 7712 idx_insert_last = i; 7713 } 7714 } 7715 } 7716 7717 // modify idx_insert from first position if needed 7718 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7719 if (idx_insert_last != -1) { 7720 idx_insert = idx_insert_last; 7721 } else { 7722 idx_insert = size; 7723 } 7724 } else { 7725 if (idx_insert_first != -1) { 7726 idx_insert = idx_insert_first + 1; 7727 } 7728 } 7729 7730 // always read samples from chain input buffer 7731 effect->setInBuffer(mInBuffer); 7732 7733 // if last effect in the chain, output samples to chain 7734 // output buffer, otherwise to chain input buffer 7735 if (idx_insert == size) { 7736 if (idx_insert != 0) { 7737 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7738 mEffects[idx_insert-1]->configure(); 7739 } 7740 effect->setOutBuffer(mOutBuffer); 7741 } else { 7742 effect->setOutBuffer(mInBuffer); 7743 } 7744 mEffects.insertAt(effect, idx_insert); 7745 7746 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7747 } 7748 effect->configure(); 7749 return NO_ERROR; 7750} 7751 7752// removeEffect_l() must be called with PlaybackThread::mLock held 7753size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7754{ 7755 Mutex::Autolock _l(mLock); 7756 size_t size = mEffects.size(); 7757 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7758 7759 for (size_t i = 0; i < size; i++) { 7760 if (effect == mEffects[i]) { 7761 // calling stop here will remove pre-processing effect from the audio HAL. 7762 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7763 // the middle of a read from audio HAL 7764 if (mEffects[i]->state() == EffectModule::ACTIVE || 7765 mEffects[i]->state() == EffectModule::STOPPING) { 7766 mEffects[i]->stop(); 7767 } 7768 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7769 delete[] effect->inBuffer(); 7770 } else { 7771 if (i == size - 1 && i != 0) { 7772 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7773 mEffects[i - 1]->configure(); 7774 } 7775 } 7776 mEffects.removeAt(i); 7777 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7778 break; 7779 } 7780 } 7781 7782 return mEffects.size(); 7783} 7784 7785// setDevice_l() must be called with PlaybackThread::mLock held 7786void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7787{ 7788 size_t size = mEffects.size(); 7789 for (size_t i = 0; i < size; i++) { 7790 mEffects[i]->setDevice(device); 7791 } 7792} 7793 7794// setMode_l() must be called with PlaybackThread::mLock held 7795void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7796{ 7797 size_t size = mEffects.size(); 7798 for (size_t i = 0; i < size; i++) { 7799 mEffects[i]->setMode(mode); 7800 } 7801} 7802 7803// setVolume_l() must be called with PlaybackThread::mLock held 7804bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7805{ 7806 uint32_t newLeft = *left; 7807 uint32_t newRight = *right; 7808 bool hasControl = false; 7809 int ctrlIdx = -1; 7810 size_t size = mEffects.size(); 7811 7812 // first update volume controller 7813 for (size_t i = size; i > 0; i--) { 7814 if (mEffects[i - 1]->isProcessEnabled() && 7815 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7816 ctrlIdx = i - 1; 7817 hasControl = true; 7818 break; 7819 } 7820 } 7821 7822 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7823 if (hasControl) { 7824 *left = mNewLeftVolume; 7825 *right = mNewRightVolume; 7826 } 7827 return hasControl; 7828 } 7829 7830 mVolumeCtrlIdx = ctrlIdx; 7831 mLeftVolume = newLeft; 7832 mRightVolume = newRight; 7833 7834 // second get volume update from volume controller 7835 if (ctrlIdx >= 0) { 7836 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7837 mNewLeftVolume = newLeft; 7838 mNewRightVolume = newRight; 7839 } 7840 // then indicate volume to all other effects in chain. 7841 // Pass altered volume to effects before volume controller 7842 // and requested volume to effects after controller 7843 uint32_t lVol = newLeft; 7844 uint32_t rVol = newRight; 7845 7846 for (size_t i = 0; i < size; i++) { 7847 if ((int)i == ctrlIdx) continue; 7848 // this also works for ctrlIdx == -1 when there is no volume controller 7849 if ((int)i > ctrlIdx) { 7850 lVol = *left; 7851 rVol = *right; 7852 } 7853 mEffects[i]->setVolume(&lVol, &rVol, false); 7854 } 7855 *left = newLeft; 7856 *right = newRight; 7857 7858 return hasControl; 7859} 7860 7861status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7862{ 7863 const size_t SIZE = 256; 7864 char buffer[SIZE]; 7865 String8 result; 7866 7867 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7868 result.append(buffer); 7869 7870 bool locked = tryLock(mLock); 7871 // failed to lock - AudioFlinger is probably deadlocked 7872 if (!locked) { 7873 result.append("\tCould not lock mutex:\n"); 7874 } 7875 7876 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7877 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7878 mEffects.size(), 7879 (uint32_t)mInBuffer, 7880 (uint32_t)mOutBuffer, 7881 mActiveTrackCnt); 7882 result.append(buffer); 7883 write(fd, result.string(), result.size()); 7884 7885 for (size_t i = 0; i < mEffects.size(); ++i) { 7886 sp<EffectModule> effect = mEffects[i]; 7887 if (effect != 0) { 7888 effect->dump(fd, args); 7889 } 7890 } 7891 7892 if (locked) { 7893 mLock.unlock(); 7894 } 7895 7896 return NO_ERROR; 7897} 7898 7899// must be called with ThreadBase::mLock held 7900void AudioFlinger::EffectChain::setEffectSuspended_l( 7901 const effect_uuid_t *type, bool suspend) 7902{ 7903 sp<SuspendedEffectDesc> desc; 7904 // use effect type UUID timelow as key as there is no real risk of identical 7905 // timeLow fields among effect type UUIDs. 7906 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7907 if (suspend) { 7908 if (index >= 0) { 7909 desc = mSuspendedEffects.valueAt(index); 7910 } else { 7911 desc = new SuspendedEffectDesc(); 7912 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7913 mSuspendedEffects.add(type->timeLow, desc); 7914 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7915 } 7916 if (desc->mRefCount++ == 0) { 7917 sp<EffectModule> effect = getEffectIfEnabled(type); 7918 if (effect != 0) { 7919 desc->mEffect = effect; 7920 effect->setSuspended(true); 7921 effect->setEnabled(false); 7922 } 7923 } 7924 } else { 7925 if (index < 0) { 7926 return; 7927 } 7928 desc = mSuspendedEffects.valueAt(index); 7929 if (desc->mRefCount <= 0) { 7930 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7931 desc->mRefCount = 1; 7932 } 7933 if (--desc->mRefCount == 0) { 7934 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7935 if (desc->mEffect != 0) { 7936 sp<EffectModule> effect = desc->mEffect.promote(); 7937 if (effect != 0) { 7938 effect->setSuspended(false); 7939 sp<EffectHandle> handle = effect->controlHandle(); 7940 if (handle != 0) { 7941 effect->setEnabled(handle->enabled()); 7942 } 7943 } 7944 desc->mEffect.clear(); 7945 } 7946 mSuspendedEffects.removeItemsAt(index); 7947 } 7948 } 7949} 7950 7951// must be called with ThreadBase::mLock held 7952void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7953{ 7954 sp<SuspendedEffectDesc> desc; 7955 7956 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7957 if (suspend) { 7958 if (index >= 0) { 7959 desc = mSuspendedEffects.valueAt(index); 7960 } else { 7961 desc = new SuspendedEffectDesc(); 7962 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7963 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7964 } 7965 if (desc->mRefCount++ == 0) { 7966 Vector< sp<EffectModule> > effects; 7967 getSuspendEligibleEffects(effects); 7968 for (size_t i = 0; i < effects.size(); i++) { 7969 setEffectSuspended_l(&effects[i]->desc().type, true); 7970 } 7971 } 7972 } else { 7973 if (index < 0) { 7974 return; 7975 } 7976 desc = mSuspendedEffects.valueAt(index); 7977 if (desc->mRefCount <= 0) { 7978 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7979 desc->mRefCount = 1; 7980 } 7981 if (--desc->mRefCount == 0) { 7982 Vector<const effect_uuid_t *> types; 7983 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7984 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7985 continue; 7986 } 7987 types.add(&mSuspendedEffects.valueAt(i)->mType); 7988 } 7989 for (size_t i = 0; i < types.size(); i++) { 7990 setEffectSuspended_l(types[i], false); 7991 } 7992 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7993 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7994 } 7995 } 7996} 7997 7998 7999// The volume effect is used for automated tests only 8000#ifndef OPENSL_ES_H_ 8001static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8002 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8003const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8004#endif //OPENSL_ES_H_ 8005 8006bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8007{ 8008 // auxiliary effects and visualizer are never suspended on output mix 8009 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8010 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8011 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8012 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8013 return false; 8014 } 8015 return true; 8016} 8017 8018void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8019{ 8020 effects.clear(); 8021 for (size_t i = 0; i < mEffects.size(); i++) { 8022 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8023 effects.add(mEffects[i]); 8024 } 8025 } 8026} 8027 8028sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8029 const effect_uuid_t *type) 8030{ 8031 sp<EffectModule> effect = getEffectFromType_l(type); 8032 return effect != 0 && effect->isEnabled() ? effect : 0; 8033} 8034 8035void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8036 bool enabled) 8037{ 8038 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8039 if (enabled) { 8040 if (index < 0) { 8041 // if the effect is not suspend check if all effects are suspended 8042 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8043 if (index < 0) { 8044 return; 8045 } 8046 if (!isEffectEligibleForSuspend(effect->desc())) { 8047 return; 8048 } 8049 setEffectSuspended_l(&effect->desc().type, enabled); 8050 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8051 if (index < 0) { 8052 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8053 return; 8054 } 8055 } 8056 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8057 effect->desc().type.timeLow); 8058 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8059 // if effect is requested to suspended but was not yet enabled, supend it now. 8060 if (desc->mEffect == 0) { 8061 desc->mEffect = effect; 8062 effect->setEnabled(false); 8063 effect->setSuspended(true); 8064 } 8065 } else { 8066 if (index < 0) { 8067 return; 8068 } 8069 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8070 effect->desc().type.timeLow); 8071 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8072 desc->mEffect.clear(); 8073 effect->setSuspended(false); 8074 } 8075} 8076 8077#undef LOG_TAG 8078#define LOG_TAG "AudioFlinger" 8079 8080// ---------------------------------------------------------------------------- 8081 8082status_t AudioFlinger::onTransact( 8083 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8084{ 8085 return BnAudioFlinger::onTransact(code, data, reply, flags); 8086} 8087 8088}; // namespace android 8089