AudioFlinger.cpp revision b279312a9038b9c5b9b05b31b1b1db86f748efd8
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 count\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->mSessionid, r->mPid, r->mCnt); 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->mPid, i); 1040 if (ref->mPid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 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, const 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 Vector< sp<Track> > tracksToRemove; 1997 1998 standbyTime = systemTime(); 1999 mixBufferSize = mFrameCount * mFrameSize; 2000 2001 // MIXER 2002 // FIXME: Relaxed timing because of a certain device that can't meet latency 2003 // Should be reduced to 2x after the vendor fixes the driver issue 2004 // increase threshold again due to low power audio mode. The way this warning threshold is 2005 // calculated and its usefulness should be reconsidered anyway. 2006 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2007 nsecs_t lastWarning = 0; 2008if (mType == MIXER) { 2009 longStandbyExit = false; 2010} 2011 2012 // DUPLICATING 2013 // FIXME could this be made local to while loop? 2014 writeFrames = 0; 2015 2016 activeSleepTime = activeSleepTimeUs(); 2017 idleSleepTime = idleSleepTimeUs(); 2018 sleepTime = idleSleepTime; 2019 2020if (mType == MIXER) { 2021 sleepTimeShift = 0; 2022} 2023 2024 // MIXER 2025 CpuStats cpuStats; 2026 2027 // DIRECT 2028if (mType == DIRECT) { 2029 // use shorter standby delay as on normal output to release 2030 // hardware resources as soon as possible 2031 standbyDelay = microseconds(activeSleepTime*2); 2032} 2033 2034 acquireWakeLock(); 2035 2036 while (!exitPending()) 2037 { 2038if (mType == MIXER) { 2039 cpuStats.sample(); 2040} 2041 2042 Vector< sp<EffectChain> > effectChains; 2043 2044 processConfigEvents(); 2045 2046if (mType == DIRECT) { 2047 activeTrack.clear(); 2048} 2049 2050 mixerStatus = MIXER_IDLE; 2051 { // scope for mLock 2052 2053 Mutex::Autolock _l(mLock); 2054 2055 if (checkForNewParameters_l()) { 2056 mixBufferSize = mFrameCount * mFrameSize; 2057 2058if (mType == MIXER) { 2059 // FIXME: Relaxed timing because of a certain device that can't meet latency 2060 // Should be reduced to 2x after the vendor fixes the driver issue 2061 // increase threshold again due to low power audio mode. The way this warning 2062 // threshold is calculated and its usefulness should be reconsidered anyway. 2063 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2064} 2065 2066 updateWaitTime_l(); 2067 2068 activeSleepTime = activeSleepTimeUs(); 2069 idleSleepTime = idleSleepTimeUs(); 2070 2071if (mType == DIRECT) { 2072 standbyDelay = microseconds(activeSleepTime*2); 2073} 2074 2075 } 2076 2077 saveOutputTracks(); 2078 2079 // put audio hardware into standby after short delay 2080 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2081 mSuspended > 0)) { 2082 if (!mStandby) { 2083 2084 threadLoop_standby(); 2085 2086 mStandby = true; 2087 mBytesWritten = 0; 2088 } 2089 2090 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2091 // we're about to wait, flush the binder command buffer 2092 IPCThreadState::self()->flushCommands(); 2093 2094 clearOutputTracks(); 2095 2096 if (exitPending()) break; 2097 2098 releaseWakeLock_l(); 2099 // wait until we have something to do... 2100 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2101 mWaitWorkCV.wait(mLock); 2102 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2103 acquireWakeLock_l(); 2104 2105if (mType == MIXER || mType == DUPLICATING) { 2106 mPrevMixerStatus = MIXER_IDLE; 2107} 2108 2109 checkSilentMode_l(); 2110 2111if (mType == MIXER || mType == DUPLICATING) { 2112 standbyTime = systemTime() + mStandbyTimeInNsecs; 2113} 2114 2115if (mType == DIRECT) { 2116 standbyTime = systemTime() + standbyDelay; 2117} 2118 2119 sleepTime = idleSleepTime; 2120 2121if (mType == MIXER) { 2122 sleepTimeShift = 0; 2123} 2124 2125 continue; 2126 } 2127 } 2128 2129 mixerStatus = prepareTracks_l(&tracksToRemove); 2130 // see FIXME in AudioFlinger.h 2131 if (mixerStatus == MIXER_CONTINUE) { 2132 continue; 2133 } 2134 2135 // prevent any changes in effect chain list and in each effect chain 2136 // during mixing and effect process as the audio buffers could be deleted 2137 // or modified if an effect is created or deleted 2138 lockEffectChains_l(effectChains); 2139 } 2140 2141if (mType == DIRECT) { 2142 // For DirectOutputThread, this test is equivalent to "activeTrack != 0" 2143} 2144 2145 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2146 threadLoop_mix(); 2147 } else { 2148 threadLoop_sleepTime(); 2149 } 2150 2151 if (mSuspended > 0) { 2152 sleepTime = suspendSleepTimeUs(); 2153 } 2154 2155 // only process effects if we're going to write 2156 if (sleepTime == 0) { 2157 for (size_t i = 0; i < effectChains.size(); i ++) { 2158 effectChains[i]->process_l(); 2159 } 2160 } 2161 2162 // enable changes in effect chain 2163 unlockEffectChains(effectChains); 2164 2165 // sleepTime == 0 means we must write to audio hardware 2166 if (sleepTime == 0) { 2167 2168 threadLoop_write(); 2169 2170if (mType == MIXER) { 2171 // write blocked detection 2172 nsecs_t now = systemTime(); 2173 nsecs_t delta = now - mLastWriteTime; 2174 if (!mStandby && delta > maxPeriod) { 2175 mNumDelayedWrites++; 2176 if ((now - lastWarning) > kWarningThrottleNs) { 2177 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2178 ns2ms(delta), mNumDelayedWrites, this); 2179 lastWarning = now; 2180 } 2181 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2182 // a different threshold. Or completely removed for what it is worth anyway... 2183 if (mStandby) { 2184 longStandbyExit = true; 2185 } 2186 } 2187} 2188 2189 mStandby = false; 2190 } else { 2191 usleep(sleepTime); 2192 } 2193 2194 // finally let go of removed track(s), without the lock held 2195 // since we can't guarantee the destructors won't acquire that 2196 // same lock. 2197 tracksToRemove.clear(); 2198 2199// FIXME merge these 2200if (mType == DIRECT) { 2201 activeTrack.clear(); 2202} 2203 // FIXME I don't understand the need for this here; 2204 // it was in the original code but maybe the 2205 // assignment in saveOutputTracks() makes this unnecessary? 2206 clearOutputTracks(); 2207 2208 // Effect chains will be actually deleted here if they were removed from 2209 // mEffectChains list during mixing or effects processing 2210 effectChains.clear(); 2211 2212 // FIXME Note that the above .clear() is no longer necessary since effectChains 2213 // is now local to this block, but will keep it for now (at least until merge done). 2214 } 2215 2216if (mType == MIXER || mType == DIRECT) { 2217 // put output stream into standby mode 2218 if (!mStandby) { 2219 mOutput->stream->common.standby(&mOutput->stream->common); 2220 } 2221} 2222if (mType == DUPLICATING) { 2223 // for DuplicatingThread, standby mode is handled by the outputTracks 2224} 2225 2226 releaseWakeLock(); 2227 2228 ALOGV("Thread %p type %d exiting", this, mType); 2229 return false; 2230} 2231 2232// shared by MIXER and DIRECT, overridden by DUPLICATING 2233void AudioFlinger::PlaybackThread::threadLoop_write() 2234{ 2235 // FIXME rewrite to reduce number of system calls 2236 mLastWriteTime = systemTime(); 2237 mInWrite = true; 2238 mBytesWritten += mixBufferSize; 2239 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2240 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2241 mNumWrites++; 2242 mInWrite = false; 2243} 2244 2245// shared by MIXER and DIRECT, overridden by DUPLICATING 2246void AudioFlinger::PlaybackThread::threadLoop_standby() 2247{ 2248 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2249 mOutput->stream->common.standby(&mOutput->stream->common); 2250} 2251 2252void AudioFlinger::MixerThread::threadLoop_mix() 2253{ 2254 // obtain the presentation timestamp of the next output buffer 2255 int64_t pts; 2256 status_t status = INVALID_OPERATION; 2257 2258 if (NULL != mOutput->stream->get_next_write_timestamp) { 2259 status = mOutput->stream->get_next_write_timestamp( 2260 mOutput->stream, &pts); 2261 } 2262 2263 if (status != NO_ERROR) { 2264 pts = AudioBufferProvider::kInvalidPTS; 2265 } 2266 2267 // mix buffers... 2268 mAudioMixer->process(pts); 2269 // increase sleep time progressively when application underrun condition clears. 2270 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2271 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2272 // such that we would underrun the audio HAL. 2273 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2274 sleepTimeShift--; 2275 } 2276 sleepTime = 0; 2277 standbyTime = systemTime() + mStandbyTimeInNsecs; 2278 //TODO: delay standby when effects have a tail 2279} 2280 2281void AudioFlinger::MixerThread::threadLoop_sleepTime() 2282{ 2283 // If no tracks are ready, sleep once for the duration of an output 2284 // buffer size, then write 0s to the output 2285 if (sleepTime == 0) { 2286 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2287 sleepTime = activeSleepTime >> sleepTimeShift; 2288 if (sleepTime < kMinThreadSleepTimeUs) { 2289 sleepTime = kMinThreadSleepTimeUs; 2290 } 2291 // reduce sleep time in case of consecutive application underruns to avoid 2292 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2293 // duration we would end up writing less data than needed by the audio HAL if 2294 // the condition persists. 2295 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2296 sleepTimeShift++; 2297 } 2298 } else { 2299 sleepTime = idleSleepTime; 2300 } 2301 } else if (mBytesWritten != 0 || 2302 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2303 memset (mMixBuffer, 0, mixBufferSize); 2304 sleepTime = 0; 2305 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2306 } 2307 // TODO add standby time extension fct of effect tail 2308} 2309 2310// prepareTracks_l() must be called with ThreadBase::mLock held 2311AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2312 Vector< sp<Track> > *tracksToRemove) 2313{ 2314 2315 mixer_state mixerStatus = MIXER_IDLE; 2316 // find out which tracks need to be processed 2317 size_t count = mActiveTracks.size(); 2318 size_t mixedTracks = 0; 2319 size_t tracksWithEffect = 0; 2320 2321 float masterVolume = mMasterVolume; 2322 bool masterMute = mMasterMute; 2323 2324 if (masterMute) { 2325 masterVolume = 0; 2326 } 2327 // Delegate master volume control to effect in output mix effect chain if needed 2328 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2329 if (chain != 0) { 2330 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2331 chain->setVolume_l(&v, &v); 2332 masterVolume = (float)((v + (1 << 23)) >> 24); 2333 chain.clear(); 2334 } 2335 2336 for (size_t i=0 ; i<count ; i++) { 2337 sp<Track> t = mActiveTracks[i].promote(); 2338 if (t == 0) continue; 2339 2340 // this const just means the local variable doesn't change 2341 Track* const track = t.get(); 2342 audio_track_cblk_t* cblk = track->cblk(); 2343 2344 // The first time a track is added we wait 2345 // for all its buffers to be filled before processing it 2346 int name = track->name(); 2347 // make sure that we have enough frames to mix one full buffer. 2348 // enforce this condition only once to enable draining the buffer in case the client 2349 // app does not call stop() and relies on underrun to stop: 2350 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2351 // during last round 2352 uint32_t minFrames = 1; 2353 if (!track->isStopped() && !track->isPausing() && 2354 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2355 if (t->sampleRate() == (int)mSampleRate) { 2356 minFrames = mFrameCount; 2357 } else { 2358 // +1 for rounding and +1 for additional sample needed for interpolation 2359 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2360 // add frames already consumed but not yet released by the resampler 2361 // because cblk->framesReady() will include these frames 2362 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2363 // the minimum track buffer size is normally twice the number of frames necessary 2364 // to fill one buffer and the resampler should not leave more than one buffer worth 2365 // of unreleased frames after each pass, but just in case... 2366 ALOG_ASSERT(minFrames <= cblk->frameCount); 2367 } 2368 } 2369 if ((track->framesReady() >= minFrames) && track->isReady() && 2370 !track->isPaused() && !track->isTerminated()) 2371 { 2372 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2373 2374 mixedTracks++; 2375 2376 // track->mainBuffer() != mMixBuffer means there is an effect chain 2377 // connected to the track 2378 chain.clear(); 2379 if (track->mainBuffer() != mMixBuffer) { 2380 chain = getEffectChain_l(track->sessionId()); 2381 // Delegate volume control to effect in track effect chain if needed 2382 if (chain != 0) { 2383 tracksWithEffect++; 2384 } else { 2385 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2386 name, track->sessionId()); 2387 } 2388 } 2389 2390 2391 int param = AudioMixer::VOLUME; 2392 if (track->mFillingUpStatus == Track::FS_FILLED) { 2393 // no ramp for the first volume setting 2394 track->mFillingUpStatus = Track::FS_ACTIVE; 2395 if (track->mState == TrackBase::RESUMING) { 2396 track->mState = TrackBase::ACTIVE; 2397 param = AudioMixer::RAMP_VOLUME; 2398 } 2399 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2400 } else if (cblk->server != 0) { 2401 // If the track is stopped before the first frame was mixed, 2402 // do not apply ramp 2403 param = AudioMixer::RAMP_VOLUME; 2404 } 2405 2406 // compute volume for this track 2407 uint32_t vl, vr, va; 2408 if (track->isMuted() || track->isPausing() || 2409 mStreamTypes[track->streamType()].mute) { 2410 vl = vr = va = 0; 2411 if (track->isPausing()) { 2412 track->setPaused(); 2413 } 2414 } else { 2415 2416 // read original volumes with volume control 2417 float typeVolume = mStreamTypes[track->streamType()].volume; 2418 float v = masterVolume * typeVolume; 2419 uint32_t vlr = cblk->getVolumeLR(); 2420 vl = vlr & 0xFFFF; 2421 vr = vlr >> 16; 2422 // track volumes come from shared memory, so can't be trusted and must be clamped 2423 if (vl > MAX_GAIN_INT) { 2424 ALOGV("Track left volume out of range: %04X", vl); 2425 vl = MAX_GAIN_INT; 2426 } 2427 if (vr > MAX_GAIN_INT) { 2428 ALOGV("Track right volume out of range: %04X", vr); 2429 vr = MAX_GAIN_INT; 2430 } 2431 // now apply the master volume and stream type volume 2432 vl = (uint32_t)(v * vl) << 12; 2433 vr = (uint32_t)(v * vr) << 12; 2434 // assuming master volume and stream type volume each go up to 1.0, 2435 // vl and vr are now in 8.24 format 2436 2437 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2438 // send level comes from shared memory and so may be corrupt 2439 if (sendLevel > MAX_GAIN_INT) { 2440 ALOGV("Track send level out of range: %04X", sendLevel); 2441 sendLevel = MAX_GAIN_INT; 2442 } 2443 va = (uint32_t)(v * sendLevel); 2444 } 2445 // Delegate volume control to effect in track effect chain if needed 2446 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2447 // Do not ramp volume if volume is controlled by effect 2448 param = AudioMixer::VOLUME; 2449 track->mHasVolumeController = true; 2450 } else { 2451 // force no volume ramp when volume controller was just disabled or removed 2452 // from effect chain to avoid volume spike 2453 if (track->mHasVolumeController) { 2454 param = AudioMixer::VOLUME; 2455 } 2456 track->mHasVolumeController = false; 2457 } 2458 2459 // Convert volumes from 8.24 to 4.12 format 2460 // This additional clamping is needed in case chain->setVolume_l() overshot 2461 vl = (vl + (1 << 11)) >> 12; 2462 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2463 vr = (vr + (1 << 11)) >> 12; 2464 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2465 2466 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2467 2468 // XXX: these things DON'T need to be done each time 2469 mAudioMixer->setBufferProvider(name, track); 2470 mAudioMixer->enable(name); 2471 2472 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2473 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2474 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2475 mAudioMixer->setParameter( 2476 name, 2477 AudioMixer::TRACK, 2478 AudioMixer::FORMAT, (void *)track->format()); 2479 mAudioMixer->setParameter( 2480 name, 2481 AudioMixer::TRACK, 2482 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2483 mAudioMixer->setParameter( 2484 name, 2485 AudioMixer::RESAMPLE, 2486 AudioMixer::SAMPLE_RATE, 2487 (void *)(cblk->sampleRate)); 2488 mAudioMixer->setParameter( 2489 name, 2490 AudioMixer::TRACK, 2491 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2492 mAudioMixer->setParameter( 2493 name, 2494 AudioMixer::TRACK, 2495 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2496 2497 // reset retry count 2498 track->mRetryCount = kMaxTrackRetries; 2499 // If one track is ready, set the mixer ready if: 2500 // - the mixer was not ready during previous round OR 2501 // - no other track is not ready 2502 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2503 mixerStatus != MIXER_TRACKS_ENABLED) { 2504 mixerStatus = MIXER_TRACKS_READY; 2505 } 2506 } else { 2507 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2508 if (track->isStopped()) { 2509 track->reset(); 2510 } 2511 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2512 // We have consumed all the buffers of this track. 2513 // Remove it from the list of active tracks. 2514 tracksToRemove->add(track); 2515 } else { 2516 // No buffers for this track. Give it a few chances to 2517 // fill a buffer, then remove it from active list. 2518 if (--(track->mRetryCount) <= 0) { 2519 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2520 tracksToRemove->add(track); 2521 // indicate to client process that the track was disabled because of underrun 2522 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2523 // If one track is not ready, mark the mixer also not ready if: 2524 // - the mixer was ready during previous round OR 2525 // - no other track is ready 2526 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2527 mixerStatus != MIXER_TRACKS_READY) { 2528 mixerStatus = MIXER_TRACKS_ENABLED; 2529 } 2530 } 2531 mAudioMixer->disable(name); 2532 } 2533 } 2534 2535 // remove all the tracks that need to be... 2536 count = tracksToRemove->size(); 2537 if (CC_UNLIKELY(count)) { 2538 for (size_t i=0 ; i<count ; i++) { 2539 const sp<Track>& track = tracksToRemove->itemAt(i); 2540 mActiveTracks.remove(track); 2541 if (track->mainBuffer() != mMixBuffer) { 2542 chain = getEffectChain_l(track->sessionId()); 2543 if (chain != 0) { 2544 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2545 chain->decActiveTrackCnt(); 2546 } 2547 } 2548 if (track->isTerminated()) { 2549 removeTrack_l(track); 2550 } 2551 } 2552 } 2553 2554 // mix buffer must be cleared if all tracks are connected to an 2555 // effect chain as in this case the mixer will not write to 2556 // mix buffer and track effects will accumulate into it 2557 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2558 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2559 } 2560 2561 mPrevMixerStatus = mixerStatus; 2562 return mixerStatus; 2563} 2564 2565void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2566{ 2567 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2568 this, streamType, mTracks.size()); 2569 Mutex::Autolock _l(mLock); 2570 2571 size_t size = mTracks.size(); 2572 for (size_t i = 0; i < size; i++) { 2573 sp<Track> t = mTracks[i]; 2574 if (t->streamType() == streamType) { 2575 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2576 t->mCblk->cv.signal(); 2577 } 2578 } 2579} 2580 2581void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2582{ 2583 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2584 this, streamType, valid); 2585 Mutex::Autolock _l(mLock); 2586 2587 mStreamTypes[streamType].valid = valid; 2588} 2589 2590// getTrackName_l() must be called with ThreadBase::mLock held 2591int AudioFlinger::MixerThread::getTrackName_l() 2592{ 2593 return mAudioMixer->getTrackName(); 2594} 2595 2596// deleteTrackName_l() must be called with ThreadBase::mLock held 2597void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2598{ 2599 ALOGV("remove track (%d) and delete from mixer", name); 2600 mAudioMixer->deleteTrackName(name); 2601} 2602 2603// checkForNewParameters_l() must be called with ThreadBase::mLock held 2604bool AudioFlinger::MixerThread::checkForNewParameters_l() 2605{ 2606 bool reconfig = false; 2607 2608 while (!mNewParameters.isEmpty()) { 2609 status_t status = NO_ERROR; 2610 String8 keyValuePair = mNewParameters[0]; 2611 AudioParameter param = AudioParameter(keyValuePair); 2612 int value; 2613 2614 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2615 reconfig = true; 2616 } 2617 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2618 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2619 status = BAD_VALUE; 2620 } else { 2621 reconfig = true; 2622 } 2623 } 2624 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2625 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2626 status = BAD_VALUE; 2627 } else { 2628 reconfig = true; 2629 } 2630 } 2631 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2632 // do not accept frame count changes if tracks are open as the track buffer 2633 // size depends on frame count and correct behavior would not be guaranteed 2634 // if frame count is changed after track creation 2635 if (!mTracks.isEmpty()) { 2636 status = INVALID_OPERATION; 2637 } else { 2638 reconfig = true; 2639 } 2640 } 2641 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2642 // when changing the audio output device, call addBatteryData to notify 2643 // the change 2644 if ((int)mDevice != value) { 2645 uint32_t params = 0; 2646 // check whether speaker is on 2647 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2648 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2649 } 2650 2651 int deviceWithoutSpeaker 2652 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2653 // check if any other device (except speaker) is on 2654 if (value & deviceWithoutSpeaker ) { 2655 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2656 } 2657 2658 if (params != 0) { 2659 addBatteryData(params); 2660 } 2661 } 2662 2663 // forward device change to effects that have requested to be 2664 // aware of attached audio device. 2665 mDevice = (uint32_t)value; 2666 for (size_t i = 0; i < mEffectChains.size(); i++) { 2667 mEffectChains[i]->setDevice_l(mDevice); 2668 } 2669 } 2670 2671 if (status == NO_ERROR) { 2672 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2673 keyValuePair.string()); 2674 if (!mStandby && status == INVALID_OPERATION) { 2675 mOutput->stream->common.standby(&mOutput->stream->common); 2676 mStandby = true; 2677 mBytesWritten = 0; 2678 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2679 keyValuePair.string()); 2680 } 2681 if (status == NO_ERROR && reconfig) { 2682 delete mAudioMixer; 2683 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2684 mAudioMixer = NULL; 2685 readOutputParameters(); 2686 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2687 for (size_t i = 0; i < mTracks.size() ; i++) { 2688 int name = getTrackName_l(); 2689 if (name < 0) break; 2690 mTracks[i]->mName = name; 2691 // limit track sample rate to 2 x new output sample rate 2692 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2693 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2694 } 2695 } 2696 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2697 } 2698 } 2699 2700 mNewParameters.removeAt(0); 2701 2702 mParamStatus = status; 2703 mParamCond.signal(); 2704 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2705 // already timed out waiting for the status and will never signal the condition. 2706 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2707 } 2708 return reconfig; 2709} 2710 2711status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2712{ 2713 const size_t SIZE = 256; 2714 char buffer[SIZE]; 2715 String8 result; 2716 2717 PlaybackThread::dumpInternals(fd, args); 2718 2719 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2720 result.append(buffer); 2721 write(fd, result.string(), result.size()); 2722 return NO_ERROR; 2723} 2724 2725uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2726{ 2727 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2728} 2729 2730uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2731{ 2732 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2733} 2734 2735// ---------------------------------------------------------------------------- 2736AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2737 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2738 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2739 // mLeftVolFloat, mRightVolFloat 2740 // mLeftVolShort, mRightVolShort 2741{ 2742} 2743 2744AudioFlinger::DirectOutputThread::~DirectOutputThread() 2745{ 2746} 2747 2748void AudioFlinger::DirectOutputThread::applyVolume() 2749{ 2750 // Do not apply volume on compressed audio 2751 if (!audio_is_linear_pcm(mFormat)) { 2752 return; 2753 } 2754 2755 // convert to signed 16 bit before volume calculation 2756 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2757 size_t count = mFrameCount * mChannelCount; 2758 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2759 int16_t *dst = mMixBuffer + count-1; 2760 while(count--) { 2761 *dst-- = (int16_t)(*src--^0x80) << 8; 2762 } 2763 } 2764 2765 size_t frameCount = mFrameCount; 2766 int16_t *out = mMixBuffer; 2767 if (rampVolume) { 2768 if (mChannelCount == 1) { 2769 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2770 int32_t vlInc = d / (int32_t)frameCount; 2771 int32_t vl = ((int32_t)mLeftVolShort << 16); 2772 do { 2773 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2774 out++; 2775 vl += vlInc; 2776 } while (--frameCount); 2777 2778 } else { 2779 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2780 int32_t vlInc = d / (int32_t)frameCount; 2781 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2782 int32_t vrInc = d / (int32_t)frameCount; 2783 int32_t vl = ((int32_t)mLeftVolShort << 16); 2784 int32_t vr = ((int32_t)mRightVolShort << 16); 2785 do { 2786 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2787 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2788 out += 2; 2789 vl += vlInc; 2790 vr += vrInc; 2791 } while (--frameCount); 2792 } 2793 } else { 2794 if (mChannelCount == 1) { 2795 do { 2796 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2797 out++; 2798 } while (--frameCount); 2799 } else { 2800 do { 2801 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2802 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2803 out += 2; 2804 } while (--frameCount); 2805 } 2806 } 2807 2808 // convert back to unsigned 8 bit after volume calculation 2809 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2810 size_t count = mFrameCount * mChannelCount; 2811 int16_t *src = mMixBuffer; 2812 uint8_t *dst = (uint8_t *)mMixBuffer; 2813 while(count--) { 2814 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2815 } 2816 } 2817 2818 mLeftVolShort = leftVol; 2819 mRightVolShort = rightVol; 2820} 2821 2822AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2823 Vector< sp<Track> > *tracksToRemove 2824) 2825{ 2826 sp<Track> trackToRemove; 2827 2828 // FIXME Temporarily renamed to avoid confusion with the member "mixerStatus" 2829 mixer_state mixerStatus_ = MIXER_IDLE; 2830 2831 // find out which tracks need to be processed 2832 if (mActiveTracks.size() != 0) { 2833 sp<Track> t = mActiveTracks[0].promote(); 2834 // see FIXME in AudioFlinger.h, return MIXER_IDLE might also work 2835 if (t == 0) return MIXER_CONTINUE; 2836 //if (t == 0) continue; 2837 2838 Track* const track = t.get(); 2839 audio_track_cblk_t* cblk = track->cblk(); 2840 2841 // The first time a track is added we wait 2842 // for all its buffers to be filled before processing it 2843 if (cblk->framesReady() && track->isReady() && 2844 !track->isPaused() && !track->isTerminated()) 2845 { 2846 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2847 2848 if (track->mFillingUpStatus == Track::FS_FILLED) { 2849 track->mFillingUpStatus = Track::FS_ACTIVE; 2850 mLeftVolFloat = mRightVolFloat = 0; 2851 mLeftVolShort = mRightVolShort = 0; 2852 if (track->mState == TrackBase::RESUMING) { 2853 track->mState = TrackBase::ACTIVE; 2854 rampVolume = true; 2855 } 2856 } else if (cblk->server != 0) { 2857 // If the track is stopped before the first frame was mixed, 2858 // do not apply ramp 2859 rampVolume = true; 2860 } 2861 // compute volume for this track 2862 float left, right; 2863 if (track->isMuted() || mMasterMute || track->isPausing() || 2864 mStreamTypes[track->streamType()].mute) { 2865 left = right = 0; 2866 if (track->isPausing()) { 2867 track->setPaused(); 2868 } 2869 } else { 2870 float typeVolume = mStreamTypes[track->streamType()].volume; 2871 float v = mMasterVolume * typeVolume; 2872 uint32_t vlr = cblk->getVolumeLR(); 2873 float v_clamped = v * (vlr & 0xFFFF); 2874 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2875 left = v_clamped/MAX_GAIN; 2876 v_clamped = v * (vlr >> 16); 2877 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2878 right = v_clamped/MAX_GAIN; 2879 } 2880 2881 if (left != mLeftVolFloat || right != mRightVolFloat) { 2882 mLeftVolFloat = left; 2883 mRightVolFloat = right; 2884 2885 // If audio HAL implements volume control, 2886 // force software volume to nominal value 2887 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2888 left = 1.0f; 2889 right = 1.0f; 2890 } 2891 2892 // Convert volumes from float to 8.24 2893 uint32_t vl = (uint32_t)(left * (1 << 24)); 2894 uint32_t vr = (uint32_t)(right * (1 << 24)); 2895 2896 // Delegate volume control to effect in track effect chain if needed 2897 // only one effect chain can be present on DirectOutputThread, so if 2898 // there is one, the track is connected to it 2899 if (!mEffectChains.isEmpty()) { 2900 // Do not ramp volume if volume is controlled by effect 2901 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2902 rampVolume = false; 2903 } 2904 } 2905 2906 // Convert volumes from 8.24 to 4.12 format 2907 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2908 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2909 leftVol = (uint16_t)v_clamped; 2910 v_clamped = (vr + (1 << 11)) >> 12; 2911 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2912 rightVol = (uint16_t)v_clamped; 2913 } else { 2914 leftVol = mLeftVolShort; 2915 rightVol = mRightVolShort; 2916 rampVolume = false; 2917 } 2918 2919 // reset retry count 2920 track->mRetryCount = kMaxTrackRetriesDirect; 2921 activeTrack = t; 2922 mixerStatus_ = MIXER_TRACKS_READY; 2923 } else { 2924 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2925 if (track->isStopped()) { 2926 track->reset(); 2927 } 2928 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2929 // We have consumed all the buffers of this track. 2930 // Remove it from the list of active tracks. 2931 trackToRemove = track; 2932 } else { 2933 // No buffers for this track. Give it a few chances to 2934 // fill a buffer, then remove it from active list. 2935 if (--(track->mRetryCount) <= 0) { 2936 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2937 trackToRemove = track; 2938 } else { 2939 mixerStatus_ = MIXER_TRACKS_ENABLED; 2940 } 2941 } 2942 } 2943 } 2944 2945 // FIXME merge this with similar code for removing multiple tracks 2946 // remove all the tracks that need to be... 2947 if (CC_UNLIKELY(trackToRemove != 0)) { 2948 tracksToRemove->add(trackToRemove); 2949 mActiveTracks.remove(trackToRemove); 2950 if (!mEffectChains.isEmpty()) { 2951 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2952 trackToRemove->sessionId()); 2953 mEffectChains[0]->decActiveTrackCnt(); 2954 } 2955 if (trackToRemove->isTerminated()) { 2956 removeTrack_l(trackToRemove); 2957 } 2958 } 2959 2960 return mixerStatus_; 2961} 2962 2963void AudioFlinger::DirectOutputThread::threadLoop_mix() 2964{ 2965 AudioBufferProvider::Buffer buffer; 2966 size_t frameCount = mFrameCount; 2967 int8_t *curBuf = (int8_t *)mMixBuffer; 2968 // output audio to hardware 2969 while (frameCount) { 2970 buffer.frameCount = frameCount; 2971 activeTrack->getNextBuffer(&buffer); 2972 if (CC_UNLIKELY(buffer.raw == NULL)) { 2973 memset(curBuf, 0, frameCount * mFrameSize); 2974 break; 2975 } 2976 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2977 frameCount -= buffer.frameCount; 2978 curBuf += buffer.frameCount * mFrameSize; 2979 activeTrack->releaseBuffer(&buffer); 2980 } 2981 sleepTime = 0; 2982 standbyTime = systemTime() + standbyDelay; 2983 applyVolume(); 2984} 2985 2986void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2987{ 2988 if (sleepTime == 0) { 2989 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2990 sleepTime = activeSleepTime; 2991 } else { 2992 sleepTime = idleSleepTime; 2993 } 2994 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2995 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2996 sleepTime = 0; 2997 } 2998} 2999 3000// getTrackName_l() must be called with ThreadBase::mLock held 3001int AudioFlinger::DirectOutputThread::getTrackName_l() 3002{ 3003 return 0; 3004} 3005 3006// deleteTrackName_l() must be called with ThreadBase::mLock held 3007void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3008{ 3009} 3010 3011// checkForNewParameters_l() must be called with ThreadBase::mLock held 3012bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3013{ 3014 bool reconfig = false; 3015 3016 while (!mNewParameters.isEmpty()) { 3017 status_t status = NO_ERROR; 3018 String8 keyValuePair = mNewParameters[0]; 3019 AudioParameter param = AudioParameter(keyValuePair); 3020 int value; 3021 3022 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3023 // do not accept frame count changes if tracks are open as the track buffer 3024 // size depends on frame count and correct behavior would not be garantied 3025 // if frame count is changed after track creation 3026 if (!mTracks.isEmpty()) { 3027 status = INVALID_OPERATION; 3028 } else { 3029 reconfig = true; 3030 } 3031 } 3032 if (status == NO_ERROR) { 3033 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3034 keyValuePair.string()); 3035 if (!mStandby && status == INVALID_OPERATION) { 3036 mOutput->stream->common.standby(&mOutput->stream->common); 3037 mStandby = true; 3038 mBytesWritten = 0; 3039 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3040 keyValuePair.string()); 3041 } 3042 if (status == NO_ERROR && reconfig) { 3043 readOutputParameters(); 3044 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3045 } 3046 } 3047 3048 mNewParameters.removeAt(0); 3049 3050 mParamStatus = status; 3051 mParamCond.signal(); 3052 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3053 // already timed out waiting for the status and will never signal the condition. 3054 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3055 } 3056 return reconfig; 3057} 3058 3059uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3060{ 3061 uint32_t time; 3062 if (audio_is_linear_pcm(mFormat)) { 3063 time = PlaybackThread::activeSleepTimeUs(); 3064 } else { 3065 time = 10000; 3066 } 3067 return time; 3068} 3069 3070uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3071{ 3072 uint32_t time; 3073 if (audio_is_linear_pcm(mFormat)) { 3074 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3075 } else { 3076 time = 10000; 3077 } 3078 return time; 3079} 3080 3081uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3082{ 3083 uint32_t time; 3084 if (audio_is_linear_pcm(mFormat)) { 3085 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3086 } else { 3087 time = 10000; 3088 } 3089 return time; 3090} 3091 3092 3093// ---------------------------------------------------------------------------- 3094 3095AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3096 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3097 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3098 mWaitTimeMs(UINT_MAX) 3099{ 3100 addOutputTrack(mainThread); 3101} 3102 3103AudioFlinger::DuplicatingThread::~DuplicatingThread() 3104{ 3105 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3106 mOutputTracks[i]->destroy(); 3107 } 3108} 3109 3110void AudioFlinger::DuplicatingThread::threadLoop_mix() 3111{ 3112 // mix buffers... 3113 if (outputsReady(outputTracks)) { 3114 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3115 } else { 3116 memset(mMixBuffer, 0, mixBufferSize); 3117 } 3118 sleepTime = 0; 3119 writeFrames = mFrameCount; 3120} 3121 3122void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3123{ 3124 if (sleepTime == 0) { 3125 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3126 sleepTime = activeSleepTime; 3127 } else { 3128 sleepTime = idleSleepTime; 3129 } 3130 } else if (mBytesWritten != 0) { 3131 // flush remaining overflow buffers in output tracks 3132 for (size_t i = 0; i < outputTracks.size(); i++) { 3133 if (outputTracks[i]->isActive()) { 3134 sleepTime = 0; 3135 writeFrames = 0; 3136 memset(mMixBuffer, 0, mixBufferSize); 3137 break; 3138 } 3139 } 3140 } 3141} 3142 3143void AudioFlinger::DuplicatingThread::threadLoop_write() 3144{ 3145 standbyTime = systemTime() + mStandbyTimeInNsecs; 3146 for (size_t i = 0; i < outputTracks.size(); i++) { 3147 outputTracks[i]->write(mMixBuffer, writeFrames); 3148 } 3149 mBytesWritten += mixBufferSize; 3150} 3151 3152void AudioFlinger::DuplicatingThread::threadLoop_standby() 3153{ 3154 // DuplicatingThread implements standby by stopping all tracks 3155 for (size_t i = 0; i < outputTracks.size(); i++) { 3156 outputTracks[i]->stop(); 3157 } 3158} 3159 3160void AudioFlinger::DuplicatingThread::saveOutputTracks() 3161{ 3162 outputTracks = mOutputTracks; 3163} 3164 3165void AudioFlinger::DuplicatingThread::clearOutputTracks() 3166{ 3167 outputTracks.clear(); 3168} 3169 3170void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3171{ 3172 Mutex::Autolock _l(mLock); 3173 // FIXME explain this formula 3174 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3175 OutputTrack *outputTrack = new OutputTrack(thread, 3176 this, 3177 mSampleRate, 3178 mFormat, 3179 mChannelMask, 3180 frameCount); 3181 if (outputTrack->cblk() != NULL) { 3182 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3183 mOutputTracks.add(outputTrack); 3184 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3185 updateWaitTime_l(); 3186 } 3187} 3188 3189void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3190{ 3191 Mutex::Autolock _l(mLock); 3192 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3193 if (mOutputTracks[i]->thread() == thread) { 3194 mOutputTracks[i]->destroy(); 3195 mOutputTracks.removeAt(i); 3196 updateWaitTime_l(); 3197 return; 3198 } 3199 } 3200 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3201} 3202 3203// caller must hold mLock 3204void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3205{ 3206 mWaitTimeMs = UINT_MAX; 3207 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3208 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3209 if (strong != 0) { 3210 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3211 if (waitTimeMs < mWaitTimeMs) { 3212 mWaitTimeMs = waitTimeMs; 3213 } 3214 } 3215 } 3216} 3217 3218 3219bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3220{ 3221 for (size_t i = 0; i < outputTracks.size(); i++) { 3222 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3223 if (thread == 0) { 3224 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3225 return false; 3226 } 3227 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3228 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3229 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3230 return false; 3231 } 3232 } 3233 return true; 3234} 3235 3236uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3237{ 3238 return (mWaitTimeMs * 1000) / 2; 3239} 3240 3241// ---------------------------------------------------------------------------- 3242 3243// TrackBase constructor must be called with AudioFlinger::mLock held 3244AudioFlinger::ThreadBase::TrackBase::TrackBase( 3245 ThreadBase *thread, 3246 const sp<Client>& client, 3247 uint32_t sampleRate, 3248 audio_format_t format, 3249 uint32_t channelMask, 3250 int frameCount, 3251 const sp<IMemory>& sharedBuffer, 3252 int sessionId) 3253 : RefBase(), 3254 mThread(thread), 3255 mClient(client), 3256 mCblk(NULL), 3257 // mBuffer 3258 // mBufferEnd 3259 mFrameCount(0), 3260 mState(IDLE), 3261 mFormat(format), 3262 mStepServerFailed(false), 3263 mSessionId(sessionId) 3264 // mChannelCount 3265 // mChannelMask 3266{ 3267 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3268 3269 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3270 size_t size = sizeof(audio_track_cblk_t); 3271 uint8_t channelCount = popcount(channelMask); 3272 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3273 if (sharedBuffer == 0) { 3274 size += bufferSize; 3275 } 3276 3277 if (client != NULL) { 3278 mCblkMemory = client->heap()->allocate(size); 3279 if (mCblkMemory != 0) { 3280 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3281 if (mCblk != NULL) { // construct the shared structure in-place. 3282 new(mCblk) audio_track_cblk_t(); 3283 // clear all buffers 3284 mCblk->frameCount = frameCount; 3285 mCblk->sampleRate = sampleRate; 3286 mChannelCount = channelCount; 3287 mChannelMask = channelMask; 3288 if (sharedBuffer == 0) { 3289 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3290 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3291 // Force underrun condition to avoid false underrun callback until first data is 3292 // written to buffer (other flags are cleared) 3293 mCblk->flags = CBLK_UNDERRUN_ON; 3294 } else { 3295 mBuffer = sharedBuffer->pointer(); 3296 } 3297 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3298 } 3299 } else { 3300 ALOGE("not enough memory for AudioTrack size=%u", size); 3301 client->heap()->dump("AudioTrack"); 3302 return; 3303 } 3304 } else { 3305 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3306 // construct the shared structure in-place. 3307 new(mCblk) audio_track_cblk_t(); 3308 // clear all buffers 3309 mCblk->frameCount = frameCount; 3310 mCblk->sampleRate = sampleRate; 3311 mChannelCount = channelCount; 3312 mChannelMask = channelMask; 3313 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3314 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3315 // Force underrun condition to avoid false underrun callback until first data is 3316 // written to buffer (other flags are cleared) 3317 mCblk->flags = CBLK_UNDERRUN_ON; 3318 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3319 } 3320} 3321 3322AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3323{ 3324 if (mCblk != NULL) { 3325 if (mClient == 0) { 3326 delete mCblk; 3327 } else { 3328 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3329 } 3330 } 3331 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3332 if (mClient != 0) { 3333 // Client destructor must run with AudioFlinger mutex locked 3334 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3335 // If the client's reference count drops to zero, the associated destructor 3336 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3337 // relying on the automatic clear() at end of scope. 3338 mClient.clear(); 3339 } 3340} 3341 3342// AudioBufferProvider interface 3343// getNextBuffer() = 0; 3344// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3345void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3346{ 3347 buffer->raw = NULL; 3348 mFrameCount = buffer->frameCount; 3349 (void) step(); // ignore return value of step() 3350 buffer->frameCount = 0; 3351} 3352 3353bool AudioFlinger::ThreadBase::TrackBase::step() { 3354 bool result; 3355 audio_track_cblk_t* cblk = this->cblk(); 3356 3357 result = cblk->stepServer(mFrameCount); 3358 if (!result) { 3359 ALOGV("stepServer failed acquiring cblk mutex"); 3360 mStepServerFailed = true; 3361 } 3362 return result; 3363} 3364 3365void AudioFlinger::ThreadBase::TrackBase::reset() { 3366 audio_track_cblk_t* cblk = this->cblk(); 3367 3368 cblk->user = 0; 3369 cblk->server = 0; 3370 cblk->userBase = 0; 3371 cblk->serverBase = 0; 3372 mStepServerFailed = false; 3373 ALOGV("TrackBase::reset"); 3374} 3375 3376int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3377 return (int)mCblk->sampleRate; 3378} 3379 3380void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3381 audio_track_cblk_t* cblk = this->cblk(); 3382 size_t frameSize = cblk->frameSize; 3383 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3384 int8_t *bufferEnd = bufferStart + frames * frameSize; 3385 3386 // Check validity of returned pointer in case the track control block would have been corrupted. 3387 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3388 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3389 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3390 server %d, serverBase %d, user %d, userBase %d", 3391 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3392 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3393 return NULL; 3394 } 3395 3396 return bufferStart; 3397} 3398 3399// ---------------------------------------------------------------------------- 3400 3401// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3402AudioFlinger::PlaybackThread::Track::Track( 3403 PlaybackThread *thread, 3404 const sp<Client>& client, 3405 audio_stream_type_t streamType, 3406 uint32_t sampleRate, 3407 audio_format_t format, 3408 uint32_t channelMask, 3409 int frameCount, 3410 const sp<IMemory>& sharedBuffer, 3411 int sessionId) 3412 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3413 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3414 mAuxEffectId(0), mHasVolumeController(false) 3415{ 3416 if (mCblk != NULL) { 3417 if (thread != NULL) { 3418 mName = thread->getTrackName_l(); 3419 mMainBuffer = thread->mixBuffer(); 3420 } 3421 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3422 if (mName < 0) { 3423 ALOGE("no more track names available"); 3424 } 3425 mStreamType = streamType; 3426 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3427 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3428 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3429 } 3430} 3431 3432AudioFlinger::PlaybackThread::Track::~Track() 3433{ 3434 ALOGV("PlaybackThread::Track destructor"); 3435 sp<ThreadBase> thread = mThread.promote(); 3436 if (thread != 0) { 3437 Mutex::Autolock _l(thread->mLock); 3438 mState = TERMINATED; 3439 } 3440} 3441 3442void AudioFlinger::PlaybackThread::Track::destroy() 3443{ 3444 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3445 // by removing it from mTracks vector, so there is a risk that this Tracks's 3446 // destructor is called. As the destructor needs to lock mLock, 3447 // we must acquire a strong reference on this Track before locking mLock 3448 // here so that the destructor is called only when exiting this function. 3449 // On the other hand, as long as Track::destroy() is only called by 3450 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3451 // this Track with its member mTrack. 3452 sp<Track> keep(this); 3453 { // scope for mLock 3454 sp<ThreadBase> thread = mThread.promote(); 3455 if (thread != 0) { 3456 if (!isOutputTrack()) { 3457 if (mState == ACTIVE || mState == RESUMING) { 3458 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3459 3460 // to track the speaker usage 3461 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3462 } 3463 AudioSystem::releaseOutput(thread->id()); 3464 } 3465 Mutex::Autolock _l(thread->mLock); 3466 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3467 playbackThread->destroyTrack_l(this); 3468 } 3469 } 3470} 3471 3472void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3473{ 3474 uint32_t vlr = mCblk->getVolumeLR(); 3475 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", 3476 mName - AudioMixer::TRACK0, 3477 (mClient == 0) ? getpid_cached : mClient->pid(), 3478 mStreamType, 3479 mFormat, 3480 mChannelMask, 3481 mSessionId, 3482 mFrameCount, 3483 mState, 3484 mMute, 3485 mFillingUpStatus, 3486 mCblk->sampleRate, 3487 vlr & 0xFFFF, 3488 vlr >> 16, 3489 mCblk->server, 3490 mCblk->user, 3491 (int)mMainBuffer, 3492 (int)mAuxBuffer); 3493} 3494 3495// AudioBufferProvider interface 3496status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3497 AudioBufferProvider::Buffer* buffer, int64_t pts) 3498{ 3499 audio_track_cblk_t* cblk = this->cblk(); 3500 uint32_t framesReady; 3501 uint32_t framesReq = buffer->frameCount; 3502 3503 // Check if last stepServer failed, try to step now 3504 if (mStepServerFailed) { 3505 if (!step()) goto getNextBuffer_exit; 3506 ALOGV("stepServer recovered"); 3507 mStepServerFailed = false; 3508 } 3509 3510 framesReady = cblk->framesReady(); 3511 3512 if (CC_LIKELY(framesReady)) { 3513 uint32_t s = cblk->server; 3514 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3515 3516 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3517 if (framesReq > framesReady) { 3518 framesReq = framesReady; 3519 } 3520 if (s + framesReq > bufferEnd) { 3521 framesReq = bufferEnd - s; 3522 } 3523 3524 buffer->raw = getBuffer(s, framesReq); 3525 if (buffer->raw == NULL) goto getNextBuffer_exit; 3526 3527 buffer->frameCount = framesReq; 3528 return NO_ERROR; 3529 } 3530 3531getNextBuffer_exit: 3532 buffer->raw = NULL; 3533 buffer->frameCount = 0; 3534 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3535 return NOT_ENOUGH_DATA; 3536} 3537 3538uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3539 return mCblk->framesReady(); 3540} 3541 3542bool AudioFlinger::PlaybackThread::Track::isReady() const { 3543 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3544 3545 if (framesReady() >= mCblk->frameCount || 3546 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3547 mFillingUpStatus = FS_FILLED; 3548 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3549 return true; 3550 } 3551 return false; 3552} 3553 3554status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3555{ 3556 status_t status = NO_ERROR; 3557 ALOGV("start(%d), calling pid %d session %d tid %d", 3558 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3559 sp<ThreadBase> thread = mThread.promote(); 3560 if (thread != 0) { 3561 Mutex::Autolock _l(thread->mLock); 3562 track_state state = mState; 3563 // here the track could be either new, or restarted 3564 // in both cases "unstop" the track 3565 if (mState == PAUSED) { 3566 mState = TrackBase::RESUMING; 3567 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3568 } else { 3569 mState = TrackBase::ACTIVE; 3570 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3571 } 3572 3573 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3574 thread->mLock.unlock(); 3575 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3576 thread->mLock.lock(); 3577 3578 // to track the speaker usage 3579 if (status == NO_ERROR) { 3580 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3581 } 3582 } 3583 if (status == NO_ERROR) { 3584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3585 playbackThread->addTrack_l(this); 3586 } else { 3587 mState = state; 3588 } 3589 } else { 3590 status = BAD_VALUE; 3591 } 3592 return status; 3593} 3594 3595void AudioFlinger::PlaybackThread::Track::stop() 3596{ 3597 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3598 sp<ThreadBase> thread = mThread.promote(); 3599 if (thread != 0) { 3600 Mutex::Autolock _l(thread->mLock); 3601 track_state state = mState; 3602 if (mState > STOPPED) { 3603 mState = STOPPED; 3604 // If the track is not active (PAUSED and buffers full), flush buffers 3605 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3606 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3607 reset(); 3608 } 3609 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3610 } 3611 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3612 thread->mLock.unlock(); 3613 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3614 thread->mLock.lock(); 3615 3616 // to track the speaker usage 3617 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3618 } 3619 } 3620} 3621 3622void AudioFlinger::PlaybackThread::Track::pause() 3623{ 3624 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3625 sp<ThreadBase> thread = mThread.promote(); 3626 if (thread != 0) { 3627 Mutex::Autolock _l(thread->mLock); 3628 if (mState == ACTIVE || mState == RESUMING) { 3629 mState = PAUSING; 3630 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3631 if (!isOutputTrack()) { 3632 thread->mLock.unlock(); 3633 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3634 thread->mLock.lock(); 3635 3636 // to track the speaker usage 3637 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3638 } 3639 } 3640 } 3641} 3642 3643void AudioFlinger::PlaybackThread::Track::flush() 3644{ 3645 ALOGV("flush(%d)", mName); 3646 sp<ThreadBase> thread = mThread.promote(); 3647 if (thread != 0) { 3648 Mutex::Autolock _l(thread->mLock); 3649 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3650 return; 3651 } 3652 // No point remaining in PAUSED state after a flush => go to 3653 // STOPPED state 3654 mState = STOPPED; 3655 3656 // do not reset the track if it is still in the process of being stopped or paused. 3657 // this will be done by prepareTracks_l() when the track is stopped. 3658 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3659 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3660 reset(); 3661 } 3662 } 3663} 3664 3665void AudioFlinger::PlaybackThread::Track::reset() 3666{ 3667 // Do not reset twice to avoid discarding data written just after a flush and before 3668 // the audioflinger thread detects the track is stopped. 3669 if (!mResetDone) { 3670 TrackBase::reset(); 3671 // Force underrun condition to avoid false underrun callback until first data is 3672 // written to buffer 3673 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3674 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3675 mFillingUpStatus = FS_FILLING; 3676 mResetDone = true; 3677 } 3678} 3679 3680void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3681{ 3682 mMute = muted; 3683} 3684 3685status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3686{ 3687 status_t status = DEAD_OBJECT; 3688 sp<ThreadBase> thread = mThread.promote(); 3689 if (thread != 0) { 3690 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3691 status = playbackThread->attachAuxEffect(this, EffectId); 3692 } 3693 return status; 3694} 3695 3696void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3697{ 3698 mAuxEffectId = EffectId; 3699 mAuxBuffer = buffer; 3700} 3701 3702// timed audio tracks 3703 3704sp<AudioFlinger::PlaybackThread::TimedTrack> 3705AudioFlinger::PlaybackThread::TimedTrack::create( 3706 PlaybackThread *thread, 3707 const sp<Client>& client, 3708 audio_stream_type_t streamType, 3709 uint32_t sampleRate, 3710 audio_format_t format, 3711 uint32_t channelMask, 3712 int frameCount, 3713 const sp<IMemory>& sharedBuffer, 3714 int sessionId) { 3715 if (!client->reserveTimedTrack()) 3716 return NULL; 3717 3718 sp<TimedTrack> track = new TimedTrack( 3719 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3720 sharedBuffer, sessionId); 3721 3722 if (track == NULL) { 3723 client->releaseTimedTrack(); 3724 return NULL; 3725 } 3726 3727 return track; 3728} 3729 3730AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3731 PlaybackThread *thread, 3732 const sp<Client>& client, 3733 audio_stream_type_t streamType, 3734 uint32_t sampleRate, 3735 audio_format_t format, 3736 uint32_t channelMask, 3737 int frameCount, 3738 const sp<IMemory>& sharedBuffer, 3739 int sessionId) 3740 : Track(thread, client, streamType, sampleRate, format, channelMask, 3741 frameCount, sharedBuffer, sessionId), 3742 mTimedSilenceBuffer(NULL), 3743 mTimedSilenceBufferSize(0), 3744 mTimedAudioOutputOnTime(false), 3745 mMediaTimeTransformValid(false) 3746{ 3747 LocalClock lc; 3748 mLocalTimeFreq = lc.getLocalFreq(); 3749 3750 mLocalTimeToSampleTransform.a_zero = 0; 3751 mLocalTimeToSampleTransform.b_zero = 0; 3752 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3753 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3754 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3755 &mLocalTimeToSampleTransform.a_to_b_denom); 3756} 3757 3758AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3759 mClient->releaseTimedTrack(); 3760 delete [] mTimedSilenceBuffer; 3761} 3762 3763status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3764 size_t size, sp<IMemory>* buffer) { 3765 3766 Mutex::Autolock _l(mTimedBufferQueueLock); 3767 3768 trimTimedBufferQueue_l(); 3769 3770 // lazily initialize the shared memory heap for timed buffers 3771 if (mTimedMemoryDealer == NULL) { 3772 const int kTimedBufferHeapSize = 512 << 10; 3773 3774 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3775 "AudioFlingerTimed"); 3776 if (mTimedMemoryDealer == NULL) 3777 return NO_MEMORY; 3778 } 3779 3780 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3781 if (newBuffer == NULL) { 3782 newBuffer = mTimedMemoryDealer->allocate(size); 3783 if (newBuffer == NULL) 3784 return NO_MEMORY; 3785 } 3786 3787 *buffer = newBuffer; 3788 return NO_ERROR; 3789} 3790 3791// caller must hold mTimedBufferQueueLock 3792void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3793 int64_t mediaTimeNow; 3794 { 3795 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3796 if (!mMediaTimeTransformValid) 3797 return; 3798 3799 int64_t targetTimeNow; 3800 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3801 ? mCCHelper.getCommonTime(&targetTimeNow) 3802 : mCCHelper.getLocalTime(&targetTimeNow); 3803 3804 if (OK != res) 3805 return; 3806 3807 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3808 &mediaTimeNow)) { 3809 return; 3810 } 3811 } 3812 3813 size_t trimIndex; 3814 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3815 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3816 break; 3817 } 3818 3819 if (trimIndex) { 3820 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3821 } 3822} 3823 3824status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3825 const sp<IMemory>& buffer, int64_t pts) { 3826 3827 { 3828 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3829 if (!mMediaTimeTransformValid) 3830 return INVALID_OPERATION; 3831 } 3832 3833 Mutex::Autolock _l(mTimedBufferQueueLock); 3834 3835 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3836 3837 return NO_ERROR; 3838} 3839 3840status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3841 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3842 3843 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3844 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3845 target); 3846 3847 if (!(target == TimedAudioTrack::LOCAL_TIME || 3848 target == TimedAudioTrack::COMMON_TIME)) { 3849 return BAD_VALUE; 3850 } 3851 3852 Mutex::Autolock lock(mMediaTimeTransformLock); 3853 mMediaTimeTransform = xform; 3854 mMediaTimeTransformTarget = target; 3855 mMediaTimeTransformValid = true; 3856 3857 return NO_ERROR; 3858} 3859 3860#define min(a, b) ((a) < (b) ? (a) : (b)) 3861 3862// implementation of getNextBuffer for tracks whose buffers have timestamps 3863status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3864 AudioBufferProvider::Buffer* buffer, int64_t pts) 3865{ 3866 if (pts == AudioBufferProvider::kInvalidPTS) { 3867 buffer->raw = 0; 3868 buffer->frameCount = 0; 3869 return INVALID_OPERATION; 3870 } 3871 3872 Mutex::Autolock _l(mTimedBufferQueueLock); 3873 3874 while (true) { 3875 3876 // if we have no timed buffers, then fail 3877 if (mTimedBufferQueue.isEmpty()) { 3878 buffer->raw = 0; 3879 buffer->frameCount = 0; 3880 return NOT_ENOUGH_DATA; 3881 } 3882 3883 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3884 3885 // calculate the PTS of the head of the timed buffer queue expressed in 3886 // local time 3887 int64_t headLocalPTS; 3888 { 3889 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3890 3891 assert(mMediaTimeTransformValid); 3892 3893 if (mMediaTimeTransform.a_to_b_denom == 0) { 3894 // the transform represents a pause, so yield silence 3895 timedYieldSilence(buffer->frameCount, buffer); 3896 return NO_ERROR; 3897 } 3898 3899 int64_t transformedPTS; 3900 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3901 &transformedPTS)) { 3902 // the transform failed. this shouldn't happen, but if it does 3903 // then just drop this buffer 3904 ALOGW("timedGetNextBuffer transform failed"); 3905 buffer->raw = 0; 3906 buffer->frameCount = 0; 3907 mTimedBufferQueue.removeAt(0); 3908 return NO_ERROR; 3909 } 3910 3911 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3912 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3913 &headLocalPTS)) { 3914 buffer->raw = 0; 3915 buffer->frameCount = 0; 3916 return INVALID_OPERATION; 3917 } 3918 } else { 3919 headLocalPTS = transformedPTS; 3920 } 3921 } 3922 3923 // adjust the head buffer's PTS to reflect the portion of the head buffer 3924 // that has already been consumed 3925 int64_t effectivePTS = headLocalPTS + 3926 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3927 3928 // Calculate the delta in samples between the head of the input buffer 3929 // queue and the start of the next output buffer that will be written. 3930 // If the transformation fails because of over or underflow, it means 3931 // that the sample's position in the output stream is so far out of 3932 // whack that it should just be dropped. 3933 int64_t sampleDelta; 3934 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3935 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3936 mTimedBufferQueue.removeAt(0); 3937 continue; 3938 } 3939 if (!mLocalTimeToSampleTransform.doForwardTransform( 3940 (effectivePTS - pts) << 32, &sampleDelta)) { 3941 ALOGV("*** too late during sample rate transform: dropped buffer"); 3942 mTimedBufferQueue.removeAt(0); 3943 continue; 3944 } 3945 3946 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3947 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3948 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3949 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3950 3951 // if the delta between the ideal placement for the next input sample and 3952 // the current output position is within this threshold, then we will 3953 // concatenate the next input samples to the previous output 3954 const int64_t kSampleContinuityThreshold = 3955 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3956 3957 // if this is the first buffer of audio that we're emitting from this track 3958 // then it should be almost exactly on time. 3959 const int64_t kSampleStartupThreshold = 1LL << 32; 3960 3961 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3962 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3963 // the next input is close enough to being on time, so concatenate it 3964 // with the last output 3965 timedYieldSamples(buffer); 3966 3967 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3968 return NO_ERROR; 3969 } else if (sampleDelta > 0) { 3970 // the gap between the current output position and the proper start of 3971 // the next input sample is too big, so fill it with silence 3972 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3973 3974 timedYieldSilence(framesUntilNextInput, buffer); 3975 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3976 return NO_ERROR; 3977 } else { 3978 // the next input sample is late 3979 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3980 size_t onTimeSamplePosition = 3981 head.position() + lateFrames * mCblk->frameSize; 3982 3983 if (onTimeSamplePosition > head.buffer()->size()) { 3984 // all the remaining samples in the head are too late, so 3985 // drop it and move on 3986 ALOGV("*** too late: dropped buffer"); 3987 mTimedBufferQueue.removeAt(0); 3988 continue; 3989 } else { 3990 // skip over the late samples 3991 head.setPosition(onTimeSamplePosition); 3992 3993 // yield the available samples 3994 timedYieldSamples(buffer); 3995 3996 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3997 return NO_ERROR; 3998 } 3999 } 4000 } 4001} 4002 4003// Yield samples from the timed buffer queue head up to the given output 4004// buffer's capacity. 4005// 4006// Caller must hold mTimedBufferQueueLock 4007void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4008 AudioBufferProvider::Buffer* buffer) { 4009 4010 const TimedBuffer& head = mTimedBufferQueue[0]; 4011 4012 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4013 head.position()); 4014 4015 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4016 mCblk->frameSize); 4017 size_t framesRequested = buffer->frameCount; 4018 buffer->frameCount = min(framesLeftInHead, framesRequested); 4019 4020 mTimedAudioOutputOnTime = true; 4021} 4022 4023// Yield samples of silence up to the given output buffer's capacity 4024// 4025// Caller must hold mTimedBufferQueueLock 4026void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4027 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4028 4029 // lazily allocate a buffer filled with silence 4030 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4031 delete [] mTimedSilenceBuffer; 4032 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4033 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4034 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4035 } 4036 4037 buffer->raw = mTimedSilenceBuffer; 4038 size_t framesRequested = buffer->frameCount; 4039 buffer->frameCount = min(numFrames, framesRequested); 4040 4041 mTimedAudioOutputOnTime = false; 4042} 4043 4044// AudioBufferProvider interface 4045void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4046 AudioBufferProvider::Buffer* buffer) { 4047 4048 Mutex::Autolock _l(mTimedBufferQueueLock); 4049 4050 // If the buffer which was just released is part of the buffer at the head 4051 // of the queue, be sure to update the amt of the buffer which has been 4052 // consumed. If the buffer being returned is not part of the head of the 4053 // queue, its either because the buffer is part of the silence buffer, or 4054 // because the head of the timed queue was trimmed after the mixer called 4055 // getNextBuffer but before the mixer called releaseBuffer. 4056 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4057 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4058 4059 void* start = head.buffer()->pointer(); 4060 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4061 4062 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4063 head.setPosition(head.position() + 4064 (buffer->frameCount * mCblk->frameSize)); 4065 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4066 mTimedBufferQueue.removeAt(0); 4067 } 4068 } 4069 } 4070 4071 buffer->raw = 0; 4072 buffer->frameCount = 0; 4073} 4074 4075uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4076 Mutex::Autolock _l(mTimedBufferQueueLock); 4077 4078 uint32_t frames = 0; 4079 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4080 const TimedBuffer& tb = mTimedBufferQueue[i]; 4081 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4082 } 4083 4084 return frames; 4085} 4086 4087AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4088 : mPTS(0), mPosition(0) {} 4089 4090AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4091 const sp<IMemory>& buffer, int64_t pts) 4092 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4093 4094// ---------------------------------------------------------------------------- 4095 4096// RecordTrack constructor must be called with AudioFlinger::mLock held 4097AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4098 RecordThread *thread, 4099 const sp<Client>& client, 4100 uint32_t sampleRate, 4101 audio_format_t format, 4102 uint32_t channelMask, 4103 int frameCount, 4104 int sessionId) 4105 : TrackBase(thread, client, sampleRate, format, 4106 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4107 mOverflow(false) 4108{ 4109 if (mCblk != NULL) { 4110 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4111 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4112 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4113 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4114 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4115 } else { 4116 mCblk->frameSize = sizeof(int8_t); 4117 } 4118 } 4119} 4120 4121AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4122{ 4123 sp<ThreadBase> thread = mThread.promote(); 4124 if (thread != 0) { 4125 AudioSystem::releaseInput(thread->id()); 4126 } 4127} 4128 4129// AudioBufferProvider interface 4130status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4131{ 4132 audio_track_cblk_t* cblk = this->cblk(); 4133 uint32_t framesAvail; 4134 uint32_t framesReq = buffer->frameCount; 4135 4136 // Check if last stepServer failed, try to step now 4137 if (mStepServerFailed) { 4138 if (!step()) goto getNextBuffer_exit; 4139 ALOGV("stepServer recovered"); 4140 mStepServerFailed = false; 4141 } 4142 4143 framesAvail = cblk->framesAvailable_l(); 4144 4145 if (CC_LIKELY(framesAvail)) { 4146 uint32_t s = cblk->server; 4147 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4148 4149 if (framesReq > framesAvail) { 4150 framesReq = framesAvail; 4151 } 4152 if (s + framesReq > bufferEnd) { 4153 framesReq = bufferEnd - s; 4154 } 4155 4156 buffer->raw = getBuffer(s, framesReq); 4157 if (buffer->raw == NULL) goto getNextBuffer_exit; 4158 4159 buffer->frameCount = framesReq; 4160 return NO_ERROR; 4161 } 4162 4163getNextBuffer_exit: 4164 buffer->raw = NULL; 4165 buffer->frameCount = 0; 4166 return NOT_ENOUGH_DATA; 4167} 4168 4169status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4170{ 4171 sp<ThreadBase> thread = mThread.promote(); 4172 if (thread != 0) { 4173 RecordThread *recordThread = (RecordThread *)thread.get(); 4174 return recordThread->start(this, tid); 4175 } else { 4176 return BAD_VALUE; 4177 } 4178} 4179 4180void AudioFlinger::RecordThread::RecordTrack::stop() 4181{ 4182 sp<ThreadBase> thread = mThread.promote(); 4183 if (thread != 0) { 4184 RecordThread *recordThread = (RecordThread *)thread.get(); 4185 recordThread->stop(this); 4186 TrackBase::reset(); 4187 // Force overerrun condition to avoid false overrun callback until first data is 4188 // read from buffer 4189 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4190 } 4191} 4192 4193void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4194{ 4195 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4196 (mClient == 0) ? getpid_cached : mClient->pid(), 4197 mFormat, 4198 mChannelMask, 4199 mSessionId, 4200 mFrameCount, 4201 mState, 4202 mCblk->sampleRate, 4203 mCblk->server, 4204 mCblk->user); 4205} 4206 4207 4208// ---------------------------------------------------------------------------- 4209 4210AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4211 PlaybackThread *playbackThread, 4212 DuplicatingThread *sourceThread, 4213 uint32_t sampleRate, 4214 audio_format_t format, 4215 uint32_t channelMask, 4216 int frameCount) 4217 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4218 mActive(false), mSourceThread(sourceThread) 4219{ 4220 4221 if (mCblk != NULL) { 4222 mCblk->flags |= CBLK_DIRECTION_OUT; 4223 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4224 mOutBuffer.frameCount = 0; 4225 playbackThread->mTracks.add(this); 4226 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4227 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4228 mCblk, mBuffer, mCblk->buffers, 4229 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4230 } else { 4231 ALOGW("Error creating output track on thread %p", playbackThread); 4232 } 4233} 4234 4235AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4236{ 4237 clearBufferQueue(); 4238} 4239 4240status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4241{ 4242 status_t status = Track::start(tid); 4243 if (status != NO_ERROR) { 4244 return status; 4245 } 4246 4247 mActive = true; 4248 mRetryCount = 127; 4249 return status; 4250} 4251 4252void AudioFlinger::PlaybackThread::OutputTrack::stop() 4253{ 4254 Track::stop(); 4255 clearBufferQueue(); 4256 mOutBuffer.frameCount = 0; 4257 mActive = false; 4258} 4259 4260bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4261{ 4262 Buffer *pInBuffer; 4263 Buffer inBuffer; 4264 uint32_t channelCount = mChannelCount; 4265 bool outputBufferFull = false; 4266 inBuffer.frameCount = frames; 4267 inBuffer.i16 = data; 4268 4269 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4270 4271 if (!mActive && frames != 0) { 4272 start(0); 4273 sp<ThreadBase> thread = mThread.promote(); 4274 if (thread != 0) { 4275 MixerThread *mixerThread = (MixerThread *)thread.get(); 4276 if (mCblk->frameCount > frames){ 4277 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4278 uint32_t startFrames = (mCblk->frameCount - frames); 4279 pInBuffer = new Buffer; 4280 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4281 pInBuffer->frameCount = startFrames; 4282 pInBuffer->i16 = pInBuffer->mBuffer; 4283 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4284 mBufferQueue.add(pInBuffer); 4285 } else { 4286 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4287 } 4288 } 4289 } 4290 } 4291 4292 while (waitTimeLeftMs) { 4293 // First write pending buffers, then new data 4294 if (mBufferQueue.size()) { 4295 pInBuffer = mBufferQueue.itemAt(0); 4296 } else { 4297 pInBuffer = &inBuffer; 4298 } 4299 4300 if (pInBuffer->frameCount == 0) { 4301 break; 4302 } 4303 4304 if (mOutBuffer.frameCount == 0) { 4305 mOutBuffer.frameCount = pInBuffer->frameCount; 4306 nsecs_t startTime = systemTime(); 4307 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4308 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4309 outputBufferFull = true; 4310 break; 4311 } 4312 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4313 if (waitTimeLeftMs >= waitTimeMs) { 4314 waitTimeLeftMs -= waitTimeMs; 4315 } else { 4316 waitTimeLeftMs = 0; 4317 } 4318 } 4319 4320 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4321 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4322 mCblk->stepUser(outFrames); 4323 pInBuffer->frameCount -= outFrames; 4324 pInBuffer->i16 += outFrames * channelCount; 4325 mOutBuffer.frameCount -= outFrames; 4326 mOutBuffer.i16 += outFrames * channelCount; 4327 4328 if (pInBuffer->frameCount == 0) { 4329 if (mBufferQueue.size()) { 4330 mBufferQueue.removeAt(0); 4331 delete [] pInBuffer->mBuffer; 4332 delete pInBuffer; 4333 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4334 } else { 4335 break; 4336 } 4337 } 4338 } 4339 4340 // If we could not write all frames, allocate a buffer and queue it for next time. 4341 if (inBuffer.frameCount) { 4342 sp<ThreadBase> thread = mThread.promote(); 4343 if (thread != 0 && !thread->standby()) { 4344 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4345 pInBuffer = new Buffer; 4346 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4347 pInBuffer->frameCount = inBuffer.frameCount; 4348 pInBuffer->i16 = pInBuffer->mBuffer; 4349 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4350 mBufferQueue.add(pInBuffer); 4351 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4352 } else { 4353 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4354 } 4355 } 4356 } 4357 4358 // Calling write() with a 0 length buffer, means that no more data will be written: 4359 // If no more buffers are pending, fill output track buffer to make sure it is started 4360 // by output mixer. 4361 if (frames == 0 && mBufferQueue.size() == 0) { 4362 if (mCblk->user < mCblk->frameCount) { 4363 frames = mCblk->frameCount - mCblk->user; 4364 pInBuffer = new Buffer; 4365 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4366 pInBuffer->frameCount = frames; 4367 pInBuffer->i16 = pInBuffer->mBuffer; 4368 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4369 mBufferQueue.add(pInBuffer); 4370 } else if (mActive) { 4371 stop(); 4372 } 4373 } 4374 4375 return outputBufferFull; 4376} 4377 4378status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4379{ 4380 int active; 4381 status_t result; 4382 audio_track_cblk_t* cblk = mCblk; 4383 uint32_t framesReq = buffer->frameCount; 4384 4385// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4386 buffer->frameCount = 0; 4387 4388 uint32_t framesAvail = cblk->framesAvailable(); 4389 4390 4391 if (framesAvail == 0) { 4392 Mutex::Autolock _l(cblk->lock); 4393 goto start_loop_here; 4394 while (framesAvail == 0) { 4395 active = mActive; 4396 if (CC_UNLIKELY(!active)) { 4397 ALOGV("Not active and NO_MORE_BUFFERS"); 4398 return NO_MORE_BUFFERS; 4399 } 4400 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4401 if (result != NO_ERROR) { 4402 return NO_MORE_BUFFERS; 4403 } 4404 // read the server count again 4405 start_loop_here: 4406 framesAvail = cblk->framesAvailable_l(); 4407 } 4408 } 4409 4410// if (framesAvail < framesReq) { 4411// return NO_MORE_BUFFERS; 4412// } 4413 4414 if (framesReq > framesAvail) { 4415 framesReq = framesAvail; 4416 } 4417 4418 uint32_t u = cblk->user; 4419 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4420 4421 if (u + framesReq > bufferEnd) { 4422 framesReq = bufferEnd - u; 4423 } 4424 4425 buffer->frameCount = framesReq; 4426 buffer->raw = (void *)cblk->buffer(u); 4427 return NO_ERROR; 4428} 4429 4430 4431void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4432{ 4433 size_t size = mBufferQueue.size(); 4434 4435 for (size_t i = 0; i < size; i++) { 4436 Buffer *pBuffer = mBufferQueue.itemAt(i); 4437 delete [] pBuffer->mBuffer; 4438 delete pBuffer; 4439 } 4440 mBufferQueue.clear(); 4441} 4442 4443// ---------------------------------------------------------------------------- 4444 4445AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4446 : RefBase(), 4447 mAudioFlinger(audioFlinger), 4448 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4449 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4450 mPid(pid), 4451 mTimedTrackCount(0) 4452{ 4453 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4454} 4455 4456// Client destructor must be called with AudioFlinger::mLock held 4457AudioFlinger::Client::~Client() 4458{ 4459 mAudioFlinger->removeClient_l(mPid); 4460} 4461 4462sp<MemoryDealer> AudioFlinger::Client::heap() const 4463{ 4464 return mMemoryDealer; 4465} 4466 4467// Reserve one of the limited slots for a timed audio track associated 4468// with this client 4469bool AudioFlinger::Client::reserveTimedTrack() 4470{ 4471 const int kMaxTimedTracksPerClient = 4; 4472 4473 Mutex::Autolock _l(mTimedTrackLock); 4474 4475 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4476 ALOGW("can not create timed track - pid %d has exceeded the limit", 4477 mPid); 4478 return false; 4479 } 4480 4481 mTimedTrackCount++; 4482 return true; 4483} 4484 4485// Release a slot for a timed audio track 4486void AudioFlinger::Client::releaseTimedTrack() 4487{ 4488 Mutex::Autolock _l(mTimedTrackLock); 4489 mTimedTrackCount--; 4490} 4491 4492// ---------------------------------------------------------------------------- 4493 4494AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4495 const sp<IAudioFlingerClient>& client, 4496 pid_t pid) 4497 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4498{ 4499} 4500 4501AudioFlinger::NotificationClient::~NotificationClient() 4502{ 4503} 4504 4505void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4506{ 4507 sp<NotificationClient> keep(this); 4508 mAudioFlinger->removeNotificationClient(mPid); 4509} 4510 4511// ---------------------------------------------------------------------------- 4512 4513AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4514 : BnAudioTrack(), 4515 mTrack(track) 4516{ 4517} 4518 4519AudioFlinger::TrackHandle::~TrackHandle() { 4520 // just stop the track on deletion, associated resources 4521 // will be freed from the main thread once all pending buffers have 4522 // been played. Unless it's not in the active track list, in which 4523 // case we free everything now... 4524 mTrack->destroy(); 4525} 4526 4527sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4528 return mTrack->getCblk(); 4529} 4530 4531status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4532 return mTrack->start(tid); 4533} 4534 4535void AudioFlinger::TrackHandle::stop() { 4536 mTrack->stop(); 4537} 4538 4539void AudioFlinger::TrackHandle::flush() { 4540 mTrack->flush(); 4541} 4542 4543void AudioFlinger::TrackHandle::mute(bool e) { 4544 mTrack->mute(e); 4545} 4546 4547void AudioFlinger::TrackHandle::pause() { 4548 mTrack->pause(); 4549} 4550 4551status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4552{ 4553 return mTrack->attachAuxEffect(EffectId); 4554} 4555 4556status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4557 sp<IMemory>* buffer) { 4558 if (!mTrack->isTimedTrack()) 4559 return INVALID_OPERATION; 4560 4561 PlaybackThread::TimedTrack* tt = 4562 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4563 return tt->allocateTimedBuffer(size, buffer); 4564} 4565 4566status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4567 int64_t pts) { 4568 if (!mTrack->isTimedTrack()) 4569 return INVALID_OPERATION; 4570 4571 PlaybackThread::TimedTrack* tt = 4572 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4573 return tt->queueTimedBuffer(buffer, pts); 4574} 4575 4576status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4577 const LinearTransform& xform, int target) { 4578 4579 if (!mTrack->isTimedTrack()) 4580 return INVALID_OPERATION; 4581 4582 PlaybackThread::TimedTrack* tt = 4583 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4584 return tt->setMediaTimeTransform( 4585 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4586} 4587 4588status_t AudioFlinger::TrackHandle::onTransact( 4589 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4590{ 4591 return BnAudioTrack::onTransact(code, data, reply, flags); 4592} 4593 4594// ---------------------------------------------------------------------------- 4595 4596sp<IAudioRecord> AudioFlinger::openRecord( 4597 pid_t pid, 4598 audio_io_handle_t input, 4599 uint32_t sampleRate, 4600 audio_format_t format, 4601 uint32_t channelMask, 4602 int frameCount, 4603 // FIXME dead, remove from IAudioFlinger 4604 uint32_t flags, 4605 int *sessionId, 4606 status_t *status) 4607{ 4608 sp<RecordThread::RecordTrack> recordTrack; 4609 sp<RecordHandle> recordHandle; 4610 sp<Client> client; 4611 status_t lStatus; 4612 RecordThread *thread; 4613 size_t inFrameCount; 4614 int lSessionId; 4615 4616 // check calling permissions 4617 if (!recordingAllowed()) { 4618 lStatus = PERMISSION_DENIED; 4619 goto Exit; 4620 } 4621 4622 // add client to list 4623 { // scope for mLock 4624 Mutex::Autolock _l(mLock); 4625 thread = checkRecordThread_l(input); 4626 if (thread == NULL) { 4627 lStatus = BAD_VALUE; 4628 goto Exit; 4629 } 4630 4631 client = registerPid_l(pid); 4632 4633 // If no audio session id is provided, create one here 4634 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4635 lSessionId = *sessionId; 4636 } else { 4637 lSessionId = nextUniqueId(); 4638 if (sessionId != NULL) { 4639 *sessionId = lSessionId; 4640 } 4641 } 4642 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4643 recordTrack = thread->createRecordTrack_l(client, 4644 sampleRate, 4645 format, 4646 channelMask, 4647 frameCount, 4648 lSessionId, 4649 &lStatus); 4650 } 4651 if (lStatus != NO_ERROR) { 4652 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4653 // destructor is called by the TrackBase destructor with mLock held 4654 client.clear(); 4655 recordTrack.clear(); 4656 goto Exit; 4657 } 4658 4659 // return to handle to client 4660 recordHandle = new RecordHandle(recordTrack); 4661 lStatus = NO_ERROR; 4662 4663Exit: 4664 if (status) { 4665 *status = lStatus; 4666 } 4667 return recordHandle; 4668} 4669 4670// ---------------------------------------------------------------------------- 4671 4672AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4673 : BnAudioRecord(), 4674 mRecordTrack(recordTrack) 4675{ 4676} 4677 4678AudioFlinger::RecordHandle::~RecordHandle() { 4679 stop(); 4680} 4681 4682sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4683 return mRecordTrack->getCblk(); 4684} 4685 4686status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4687 ALOGV("RecordHandle::start()"); 4688 return mRecordTrack->start(tid); 4689} 4690 4691void AudioFlinger::RecordHandle::stop() { 4692 ALOGV("RecordHandle::stop()"); 4693 mRecordTrack->stop(); 4694} 4695 4696status_t AudioFlinger::RecordHandle::onTransact( 4697 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4698{ 4699 return BnAudioRecord::onTransact(code, data, reply, flags); 4700} 4701 4702// ---------------------------------------------------------------------------- 4703 4704AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4705 AudioStreamIn *input, 4706 uint32_t sampleRate, 4707 uint32_t channels, 4708 audio_io_handle_t id, 4709 uint32_t device) : 4710 ThreadBase(audioFlinger, id, device, RECORD), 4711 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4712 // mRsmpInIndex and mInputBytes set by readInputParameters() 4713 mReqChannelCount(popcount(channels)), 4714 mReqSampleRate(sampleRate) 4715 // mBytesRead is only meaningful while active, and so is cleared in start() 4716 // (but might be better to also clear here for dump?) 4717{ 4718 snprintf(mName, kNameLength, "AudioIn_%X", id); 4719 4720 readInputParameters(); 4721} 4722 4723 4724AudioFlinger::RecordThread::~RecordThread() 4725{ 4726 delete[] mRsmpInBuffer; 4727 delete mResampler; 4728 delete[] mRsmpOutBuffer; 4729} 4730 4731void AudioFlinger::RecordThread::onFirstRef() 4732{ 4733 run(mName, PRIORITY_URGENT_AUDIO); 4734} 4735 4736status_t AudioFlinger::RecordThread::readyToRun() 4737{ 4738 status_t status = initCheck(); 4739 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4740 return status; 4741} 4742 4743bool AudioFlinger::RecordThread::threadLoop() 4744{ 4745 AudioBufferProvider::Buffer buffer; 4746 sp<RecordTrack> activeTrack; 4747 Vector< sp<EffectChain> > effectChains; 4748 4749 nsecs_t lastWarning = 0; 4750 4751 acquireWakeLock(); 4752 4753 // start recording 4754 while (!exitPending()) { 4755 4756 processConfigEvents(); 4757 4758 { // scope for mLock 4759 Mutex::Autolock _l(mLock); 4760 checkForNewParameters_l(); 4761 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4762 if (!mStandby) { 4763 mInput->stream->common.standby(&mInput->stream->common); 4764 mStandby = true; 4765 } 4766 4767 if (exitPending()) break; 4768 4769 releaseWakeLock_l(); 4770 ALOGV("RecordThread: loop stopping"); 4771 // go to sleep 4772 mWaitWorkCV.wait(mLock); 4773 ALOGV("RecordThread: loop starting"); 4774 acquireWakeLock_l(); 4775 continue; 4776 } 4777 if (mActiveTrack != 0) { 4778 if (mActiveTrack->mState == TrackBase::PAUSING) { 4779 if (!mStandby) { 4780 mInput->stream->common.standby(&mInput->stream->common); 4781 mStandby = true; 4782 } 4783 mActiveTrack.clear(); 4784 mStartStopCond.broadcast(); 4785 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4786 if (mReqChannelCount != mActiveTrack->channelCount()) { 4787 mActiveTrack.clear(); 4788 mStartStopCond.broadcast(); 4789 } else if (mBytesRead != 0) { 4790 // record start succeeds only if first read from audio input 4791 // succeeds 4792 if (mBytesRead > 0) { 4793 mActiveTrack->mState = TrackBase::ACTIVE; 4794 } else { 4795 mActiveTrack.clear(); 4796 } 4797 mStartStopCond.broadcast(); 4798 } 4799 mStandby = false; 4800 } 4801 } 4802 lockEffectChains_l(effectChains); 4803 } 4804 4805 if (mActiveTrack != 0) { 4806 if (mActiveTrack->mState != TrackBase::ACTIVE && 4807 mActiveTrack->mState != TrackBase::RESUMING) { 4808 unlockEffectChains(effectChains); 4809 usleep(kRecordThreadSleepUs); 4810 continue; 4811 } 4812 for (size_t i = 0; i < effectChains.size(); i ++) { 4813 effectChains[i]->process_l(); 4814 } 4815 4816 buffer.frameCount = mFrameCount; 4817 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4818 size_t framesOut = buffer.frameCount; 4819 if (mResampler == NULL) { 4820 // no resampling 4821 while (framesOut) { 4822 size_t framesIn = mFrameCount - mRsmpInIndex; 4823 if (framesIn) { 4824 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4825 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4826 if (framesIn > framesOut) 4827 framesIn = framesOut; 4828 mRsmpInIndex += framesIn; 4829 framesOut -= framesIn; 4830 if ((int)mChannelCount == mReqChannelCount || 4831 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4832 memcpy(dst, src, framesIn * mFrameSize); 4833 } else { 4834 int16_t *src16 = (int16_t *)src; 4835 int16_t *dst16 = (int16_t *)dst; 4836 if (mChannelCount == 1) { 4837 while (framesIn--) { 4838 *dst16++ = *src16; 4839 *dst16++ = *src16++; 4840 } 4841 } else { 4842 while (framesIn--) { 4843 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4844 src16 += 2; 4845 } 4846 } 4847 } 4848 } 4849 if (framesOut && mFrameCount == mRsmpInIndex) { 4850 if (framesOut == mFrameCount && 4851 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4852 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4853 framesOut = 0; 4854 } else { 4855 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4856 mRsmpInIndex = 0; 4857 } 4858 if (mBytesRead < 0) { 4859 ALOGE("Error reading audio input"); 4860 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4861 // Force input into standby so that it tries to 4862 // recover at next read attempt 4863 mInput->stream->common.standby(&mInput->stream->common); 4864 usleep(kRecordThreadSleepUs); 4865 } 4866 mRsmpInIndex = mFrameCount; 4867 framesOut = 0; 4868 buffer.frameCount = 0; 4869 } 4870 } 4871 } 4872 } else { 4873 // resampling 4874 4875 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4876 // alter output frame count as if we were expecting stereo samples 4877 if (mChannelCount == 1 && mReqChannelCount == 1) { 4878 framesOut >>= 1; 4879 } 4880 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4881 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4882 // are 32 bit aligned which should be always true. 4883 if (mChannelCount == 2 && mReqChannelCount == 1) { 4884 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4885 // the resampler always outputs stereo samples: do post stereo to mono conversion 4886 int16_t *src = (int16_t *)mRsmpOutBuffer; 4887 int16_t *dst = buffer.i16; 4888 while (framesOut--) { 4889 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4890 src += 2; 4891 } 4892 } else { 4893 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4894 } 4895 4896 } 4897 mActiveTrack->releaseBuffer(&buffer); 4898 mActiveTrack->overflow(); 4899 } 4900 // client isn't retrieving buffers fast enough 4901 else { 4902 if (!mActiveTrack->setOverflow()) { 4903 nsecs_t now = systemTime(); 4904 if ((now - lastWarning) > kWarningThrottleNs) { 4905 ALOGW("RecordThread: buffer overflow"); 4906 lastWarning = now; 4907 } 4908 } 4909 // Release the processor for a while before asking for a new buffer. 4910 // This will give the application more chance to read from the buffer and 4911 // clear the overflow. 4912 usleep(kRecordThreadSleepUs); 4913 } 4914 } 4915 // enable changes in effect chain 4916 unlockEffectChains(effectChains); 4917 effectChains.clear(); 4918 } 4919 4920 if (!mStandby) { 4921 mInput->stream->common.standby(&mInput->stream->common); 4922 } 4923 mActiveTrack.clear(); 4924 4925 mStartStopCond.broadcast(); 4926 4927 releaseWakeLock(); 4928 4929 ALOGV("RecordThread %p exiting", this); 4930 return false; 4931} 4932 4933 4934sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4935 const sp<AudioFlinger::Client>& client, 4936 uint32_t sampleRate, 4937 audio_format_t format, 4938 int channelMask, 4939 int frameCount, 4940 int sessionId, 4941 status_t *status) 4942{ 4943 sp<RecordTrack> track; 4944 status_t lStatus; 4945 4946 lStatus = initCheck(); 4947 if (lStatus != NO_ERROR) { 4948 ALOGE("Audio driver not initialized."); 4949 goto Exit; 4950 } 4951 4952 { // scope for mLock 4953 Mutex::Autolock _l(mLock); 4954 4955 track = new RecordTrack(this, client, sampleRate, 4956 format, channelMask, frameCount, sessionId); 4957 4958 if (track->getCblk() == 0) { 4959 lStatus = NO_MEMORY; 4960 goto Exit; 4961 } 4962 4963 mTrack = track.get(); 4964 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4965 bool suspend = audio_is_bluetooth_sco_device( 4966 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4967 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4968 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4969 } 4970 lStatus = NO_ERROR; 4971 4972Exit: 4973 if (status) { 4974 *status = lStatus; 4975 } 4976 return track; 4977} 4978 4979status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4980{ 4981 ALOGV("RecordThread::start tid=%d", tid); 4982 sp <ThreadBase> strongMe = this; 4983 status_t status = NO_ERROR; 4984 { 4985 AutoMutex lock(mLock); 4986 if (mActiveTrack != 0) { 4987 if (recordTrack != mActiveTrack.get()) { 4988 status = -EBUSY; 4989 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4990 mActiveTrack->mState = TrackBase::ACTIVE; 4991 } 4992 return status; 4993 } 4994 4995 recordTrack->mState = TrackBase::IDLE; 4996 mActiveTrack = recordTrack; 4997 mLock.unlock(); 4998 status_t status = AudioSystem::startInput(mId); 4999 mLock.lock(); 5000 if (status != NO_ERROR) { 5001 mActiveTrack.clear(); 5002 return status; 5003 } 5004 mRsmpInIndex = mFrameCount; 5005 mBytesRead = 0; 5006 if (mResampler != NULL) { 5007 mResampler->reset(); 5008 } 5009 mActiveTrack->mState = TrackBase::RESUMING; 5010 // signal thread to start 5011 ALOGV("Signal record thread"); 5012 mWaitWorkCV.signal(); 5013 // do not wait for mStartStopCond if exiting 5014 if (exitPending()) { 5015 mActiveTrack.clear(); 5016 status = INVALID_OPERATION; 5017 goto startError; 5018 } 5019 mStartStopCond.wait(mLock); 5020 if (mActiveTrack == 0) { 5021 ALOGV("Record failed to start"); 5022 status = BAD_VALUE; 5023 goto startError; 5024 } 5025 ALOGV("Record started OK"); 5026 return status; 5027 } 5028startError: 5029 AudioSystem::stopInput(mId); 5030 return status; 5031} 5032 5033void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5034 ALOGV("RecordThread::stop"); 5035 sp <ThreadBase> strongMe = this; 5036 { 5037 AutoMutex lock(mLock); 5038 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5039 mActiveTrack->mState = TrackBase::PAUSING; 5040 // do not wait for mStartStopCond if exiting 5041 if (exitPending()) { 5042 return; 5043 } 5044 mStartStopCond.wait(mLock); 5045 // if we have been restarted, recordTrack == mActiveTrack.get() here 5046 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5047 mLock.unlock(); 5048 AudioSystem::stopInput(mId); 5049 mLock.lock(); 5050 ALOGV("Record stopped OK"); 5051 } 5052 } 5053 } 5054} 5055 5056status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5057{ 5058 const size_t SIZE = 256; 5059 char buffer[SIZE]; 5060 String8 result; 5061 5062 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5063 result.append(buffer); 5064 5065 if (mActiveTrack != 0) { 5066 result.append("Active Track:\n"); 5067 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5068 mActiveTrack->dump(buffer, SIZE); 5069 result.append(buffer); 5070 5071 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5072 result.append(buffer); 5073 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5074 result.append(buffer); 5075 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5076 result.append(buffer); 5077 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5078 result.append(buffer); 5079 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5080 result.append(buffer); 5081 5082 5083 } else { 5084 result.append("No record client\n"); 5085 } 5086 write(fd, result.string(), result.size()); 5087 5088 dumpBase(fd, args); 5089 dumpEffectChains(fd, args); 5090 5091 return NO_ERROR; 5092} 5093 5094// AudioBufferProvider interface 5095status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5096{ 5097 size_t framesReq = buffer->frameCount; 5098 size_t framesReady = mFrameCount - mRsmpInIndex; 5099 int channelCount; 5100 5101 if (framesReady == 0) { 5102 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5103 if (mBytesRead < 0) { 5104 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5105 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5106 // Force input into standby so that it tries to 5107 // recover at next read attempt 5108 mInput->stream->common.standby(&mInput->stream->common); 5109 usleep(kRecordThreadSleepUs); 5110 } 5111 buffer->raw = NULL; 5112 buffer->frameCount = 0; 5113 return NOT_ENOUGH_DATA; 5114 } 5115 mRsmpInIndex = 0; 5116 framesReady = mFrameCount; 5117 } 5118 5119 if (framesReq > framesReady) { 5120 framesReq = framesReady; 5121 } 5122 5123 if (mChannelCount == 1 && mReqChannelCount == 2) { 5124 channelCount = 1; 5125 } else { 5126 channelCount = 2; 5127 } 5128 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5129 buffer->frameCount = framesReq; 5130 return NO_ERROR; 5131} 5132 5133// AudioBufferProvider interface 5134void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5135{ 5136 mRsmpInIndex += buffer->frameCount; 5137 buffer->frameCount = 0; 5138} 5139 5140bool AudioFlinger::RecordThread::checkForNewParameters_l() 5141{ 5142 bool reconfig = false; 5143 5144 while (!mNewParameters.isEmpty()) { 5145 status_t status = NO_ERROR; 5146 String8 keyValuePair = mNewParameters[0]; 5147 AudioParameter param = AudioParameter(keyValuePair); 5148 int value; 5149 audio_format_t reqFormat = mFormat; 5150 int reqSamplingRate = mReqSampleRate; 5151 int reqChannelCount = mReqChannelCount; 5152 5153 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5154 reqSamplingRate = value; 5155 reconfig = true; 5156 } 5157 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5158 reqFormat = (audio_format_t) value; 5159 reconfig = true; 5160 } 5161 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5162 reqChannelCount = popcount(value); 5163 reconfig = true; 5164 } 5165 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5166 // do not accept frame count changes if tracks are open as the track buffer 5167 // size depends on frame count and correct behavior would not be guaranteed 5168 // if frame count is changed after track creation 5169 if (mActiveTrack != 0) { 5170 status = INVALID_OPERATION; 5171 } else { 5172 reconfig = true; 5173 } 5174 } 5175 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5176 // forward device change to effects that have requested to be 5177 // aware of attached audio device. 5178 for (size_t i = 0; i < mEffectChains.size(); i++) { 5179 mEffectChains[i]->setDevice_l(value); 5180 } 5181 // store input device and output device but do not forward output device to audio HAL. 5182 // Note that status is ignored by the caller for output device 5183 // (see AudioFlinger::setParameters() 5184 if (value & AUDIO_DEVICE_OUT_ALL) { 5185 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5186 status = BAD_VALUE; 5187 } else { 5188 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5189 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5190 if (mTrack != NULL) { 5191 bool suspend = audio_is_bluetooth_sco_device( 5192 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5193 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5194 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5195 } 5196 } 5197 mDevice |= (uint32_t)value; 5198 } 5199 if (status == NO_ERROR) { 5200 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5201 if (status == INVALID_OPERATION) { 5202 mInput->stream->common.standby(&mInput->stream->common); 5203 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5204 } 5205 if (reconfig) { 5206 if (status == BAD_VALUE && 5207 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5208 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5209 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5210 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5211 (reqChannelCount < 3)) { 5212 status = NO_ERROR; 5213 } 5214 if (status == NO_ERROR) { 5215 readInputParameters(); 5216 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5217 } 5218 } 5219 } 5220 5221 mNewParameters.removeAt(0); 5222 5223 mParamStatus = status; 5224 mParamCond.signal(); 5225 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5226 // already timed out waiting for the status and will never signal the condition. 5227 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5228 } 5229 return reconfig; 5230} 5231 5232String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5233{ 5234 char *s; 5235 String8 out_s8 = String8(); 5236 5237 Mutex::Autolock _l(mLock); 5238 if (initCheck() != NO_ERROR) { 5239 return out_s8; 5240 } 5241 5242 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5243 out_s8 = String8(s); 5244 free(s); 5245 return out_s8; 5246} 5247 5248void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5249 AudioSystem::OutputDescriptor desc; 5250 void *param2 = NULL; 5251 5252 switch (event) { 5253 case AudioSystem::INPUT_OPENED: 5254 case AudioSystem::INPUT_CONFIG_CHANGED: 5255 desc.channels = mChannelMask; 5256 desc.samplingRate = mSampleRate; 5257 desc.format = mFormat; 5258 desc.frameCount = mFrameCount; 5259 desc.latency = 0; 5260 param2 = &desc; 5261 break; 5262 5263 case AudioSystem::INPUT_CLOSED: 5264 default: 5265 break; 5266 } 5267 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5268} 5269 5270void AudioFlinger::RecordThread::readInputParameters() 5271{ 5272 delete mRsmpInBuffer; 5273 // mRsmpInBuffer is always assigned a new[] below 5274 delete mRsmpOutBuffer; 5275 mRsmpOutBuffer = NULL; 5276 delete mResampler; 5277 mResampler = NULL; 5278 5279 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5280 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5281 mChannelCount = (uint16_t)popcount(mChannelMask); 5282 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5283 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5284 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5285 mFrameCount = mInputBytes / mFrameSize; 5286 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5287 5288 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5289 { 5290 int channelCount; 5291 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5292 // stereo to mono post process as the resampler always outputs stereo. 5293 if (mChannelCount == 1 && mReqChannelCount == 2) { 5294 channelCount = 1; 5295 } else { 5296 channelCount = 2; 5297 } 5298 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5299 mResampler->setSampleRate(mSampleRate); 5300 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5301 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5302 5303 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5304 if (mChannelCount == 1 && mReqChannelCount == 1) { 5305 mFrameCount >>= 1; 5306 } 5307 5308 } 5309 mRsmpInIndex = mFrameCount; 5310} 5311 5312unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5313{ 5314 Mutex::Autolock _l(mLock); 5315 if (initCheck() != NO_ERROR) { 5316 return 0; 5317 } 5318 5319 return mInput->stream->get_input_frames_lost(mInput->stream); 5320} 5321 5322uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5323{ 5324 Mutex::Autolock _l(mLock); 5325 uint32_t result = 0; 5326 if (getEffectChain_l(sessionId) != 0) { 5327 result = EFFECT_SESSION; 5328 } 5329 5330 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5331 result |= TRACK_SESSION; 5332 } 5333 5334 return result; 5335} 5336 5337AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5338{ 5339 Mutex::Autolock _l(mLock); 5340 return mTrack; 5341} 5342 5343AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5344{ 5345 Mutex::Autolock _l(mLock); 5346 return mInput; 5347} 5348 5349AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5350{ 5351 Mutex::Autolock _l(mLock); 5352 AudioStreamIn *input = mInput; 5353 mInput = NULL; 5354 return input; 5355} 5356 5357// this method must always be called either with ThreadBase mLock held or inside the thread loop 5358audio_stream_t* AudioFlinger::RecordThread::stream() 5359{ 5360 if (mInput == NULL) { 5361 return NULL; 5362 } 5363 return &mInput->stream->common; 5364} 5365 5366 5367// ---------------------------------------------------------------------------- 5368 5369audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5370 uint32_t *pSamplingRate, 5371 audio_format_t *pFormat, 5372 uint32_t *pChannels, 5373 uint32_t *pLatencyMs, 5374 uint32_t flags) 5375{ 5376 status_t status; 5377 PlaybackThread *thread = NULL; 5378 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5379 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5380 uint32_t channels = pChannels ? *pChannels : 0; 5381 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5382 audio_stream_out_t *outStream; 5383 audio_hw_device_t *outHwDev; 5384 5385 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5386 pDevices ? *pDevices : 0, 5387 samplingRate, 5388 format, 5389 channels, 5390 flags); 5391 5392 if (pDevices == NULL || *pDevices == 0) { 5393 return 0; 5394 } 5395 5396 Mutex::Autolock _l(mLock); 5397 5398 outHwDev = findSuitableHwDev_l(*pDevices); 5399 if (outHwDev == NULL) 5400 return 0; 5401 5402 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5403 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5404 &channels, &samplingRate, &outStream); 5405 mHardwareStatus = AUDIO_HW_IDLE; 5406 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5407 outStream, 5408 samplingRate, 5409 format, 5410 channels, 5411 status); 5412 5413 if (outStream != NULL) { 5414 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5415 audio_io_handle_t id = nextUniqueId(); 5416 5417 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5418 (format != AUDIO_FORMAT_PCM_16_BIT) || 5419 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5420 thread = new DirectOutputThread(this, output, id, *pDevices); 5421 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5422 } else { 5423 thread = new MixerThread(this, output, id, *pDevices); 5424 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5425 } 5426 mPlaybackThreads.add(id, thread); 5427 5428 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5429 if (pFormat != NULL) *pFormat = format; 5430 if (pChannels != NULL) *pChannels = channels; 5431 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5432 5433 // notify client processes of the new output creation 5434 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5435 return id; 5436 } 5437 5438 return 0; 5439} 5440 5441audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5442 audio_io_handle_t output2) 5443{ 5444 Mutex::Autolock _l(mLock); 5445 MixerThread *thread1 = checkMixerThread_l(output1); 5446 MixerThread *thread2 = checkMixerThread_l(output2); 5447 5448 if (thread1 == NULL || thread2 == NULL) { 5449 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5450 return 0; 5451 } 5452 5453 audio_io_handle_t id = nextUniqueId(); 5454 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5455 thread->addOutputTrack(thread2); 5456 mPlaybackThreads.add(id, thread); 5457 // notify client processes of the new output creation 5458 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5459 return id; 5460} 5461 5462status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5463{ 5464 // keep strong reference on the playback thread so that 5465 // it is not destroyed while exit() is executed 5466 sp <PlaybackThread> thread; 5467 { 5468 Mutex::Autolock _l(mLock); 5469 thread = checkPlaybackThread_l(output); 5470 if (thread == NULL) { 5471 return BAD_VALUE; 5472 } 5473 5474 ALOGV("closeOutput() %d", output); 5475 5476 if (thread->type() == ThreadBase::MIXER) { 5477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5478 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5479 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5480 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5481 } 5482 } 5483 } 5484 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5485 mPlaybackThreads.removeItem(output); 5486 } 5487 thread->exit(); 5488 // The thread entity (active unit of execution) is no longer running here, 5489 // but the ThreadBase container still exists. 5490 5491 if (thread->type() != ThreadBase::DUPLICATING) { 5492 AudioStreamOut *out = thread->clearOutput(); 5493 assert(out != NULL); 5494 // from now on thread->mOutput is NULL 5495 out->hwDev->close_output_stream(out->hwDev, out->stream); 5496 delete out; 5497 } 5498 return NO_ERROR; 5499} 5500 5501status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5502{ 5503 Mutex::Autolock _l(mLock); 5504 PlaybackThread *thread = checkPlaybackThread_l(output); 5505 5506 if (thread == NULL) { 5507 return BAD_VALUE; 5508 } 5509 5510 ALOGV("suspendOutput() %d", output); 5511 thread->suspend(); 5512 5513 return NO_ERROR; 5514} 5515 5516status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5517{ 5518 Mutex::Autolock _l(mLock); 5519 PlaybackThread *thread = checkPlaybackThread_l(output); 5520 5521 if (thread == NULL) { 5522 return BAD_VALUE; 5523 } 5524 5525 ALOGV("restoreOutput() %d", output); 5526 5527 thread->restore(); 5528 5529 return NO_ERROR; 5530} 5531 5532audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5533 uint32_t *pSamplingRate, 5534 audio_format_t *pFormat, 5535 uint32_t *pChannels, 5536 audio_in_acoustics_t acoustics) 5537{ 5538 status_t status; 5539 RecordThread *thread = NULL; 5540 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5541 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5542 uint32_t channels = pChannels ? *pChannels : 0; 5543 uint32_t reqSamplingRate = samplingRate; 5544 audio_format_t reqFormat = format; 5545 uint32_t reqChannels = channels; 5546 audio_stream_in_t *inStream; 5547 audio_hw_device_t *inHwDev; 5548 5549 if (pDevices == NULL || *pDevices == 0) { 5550 return 0; 5551 } 5552 5553 Mutex::Autolock _l(mLock); 5554 5555 inHwDev = findSuitableHwDev_l(*pDevices); 5556 if (inHwDev == NULL) 5557 return 0; 5558 5559 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5560 &channels, &samplingRate, 5561 acoustics, 5562 &inStream); 5563 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5564 inStream, 5565 samplingRate, 5566 format, 5567 channels, 5568 acoustics, 5569 status); 5570 5571 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5572 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5573 // or stereo to mono conversions on 16 bit PCM inputs. 5574 if (inStream == NULL && status == BAD_VALUE && 5575 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5576 (samplingRate <= 2 * reqSamplingRate) && 5577 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5578 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5579 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5580 &channels, &samplingRate, 5581 acoustics, 5582 &inStream); 5583 } 5584 5585 if (inStream != NULL) { 5586 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5587 5588 audio_io_handle_t id = nextUniqueId(); 5589 // Start record thread 5590 // RecorThread require both input and output device indication to forward to audio 5591 // pre processing modules 5592 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5593 thread = new RecordThread(this, 5594 input, 5595 reqSamplingRate, 5596 reqChannels, 5597 id, 5598 device); 5599 mRecordThreads.add(id, thread); 5600 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5601 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5602 if (pFormat != NULL) *pFormat = format; 5603 if (pChannels != NULL) *pChannels = reqChannels; 5604 5605 input->stream->common.standby(&input->stream->common); 5606 5607 // notify client processes of the new input creation 5608 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5609 return id; 5610 } 5611 5612 return 0; 5613} 5614 5615status_t AudioFlinger::closeInput(audio_io_handle_t input) 5616{ 5617 // keep strong reference on the record thread so that 5618 // it is not destroyed while exit() is executed 5619 sp <RecordThread> thread; 5620 { 5621 Mutex::Autolock _l(mLock); 5622 thread = checkRecordThread_l(input); 5623 if (thread == NULL) { 5624 return BAD_VALUE; 5625 } 5626 5627 ALOGV("closeInput() %d", input); 5628 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5629 mRecordThreads.removeItem(input); 5630 } 5631 thread->exit(); 5632 // The thread entity (active unit of execution) is no longer running here, 5633 // but the ThreadBase container still exists. 5634 5635 AudioStreamIn *in = thread->clearInput(); 5636 assert(in != NULL); 5637 // from now on thread->mInput is NULL 5638 in->hwDev->close_input_stream(in->hwDev, in->stream); 5639 delete in; 5640 5641 return NO_ERROR; 5642} 5643 5644status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5645{ 5646 Mutex::Autolock _l(mLock); 5647 MixerThread *dstThread = checkMixerThread_l(output); 5648 if (dstThread == NULL) { 5649 ALOGW("setStreamOutput() bad output id %d", output); 5650 return BAD_VALUE; 5651 } 5652 5653 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5654 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5655 5656 dstThread->setStreamValid(stream, true); 5657 5658 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5659 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5660 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5661 MixerThread *srcThread = (MixerThread *)thread; 5662 srcThread->setStreamValid(stream, false); 5663 srcThread->invalidateTracks(stream); 5664 } 5665 } 5666 5667 return NO_ERROR; 5668} 5669 5670 5671int AudioFlinger::newAudioSessionId() 5672{ 5673 return nextUniqueId(); 5674} 5675 5676void AudioFlinger::acquireAudioSessionId(int audioSession) 5677{ 5678 Mutex::Autolock _l(mLock); 5679 pid_t caller = IPCThreadState::self()->getCallingPid(); 5680 ALOGV("acquiring %d from %d", audioSession, caller); 5681 size_t num = mAudioSessionRefs.size(); 5682 for (size_t i = 0; i< num; i++) { 5683 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5684 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5685 ref->mCnt++; 5686 ALOGV(" incremented refcount to %d", ref->mCnt); 5687 return; 5688 } 5689 } 5690 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5691 ALOGV(" added new entry for %d", audioSession); 5692} 5693 5694void AudioFlinger::releaseAudioSessionId(int audioSession) 5695{ 5696 Mutex::Autolock _l(mLock); 5697 pid_t caller = IPCThreadState::self()->getCallingPid(); 5698 ALOGV("releasing %d from %d", audioSession, caller); 5699 size_t num = mAudioSessionRefs.size(); 5700 for (size_t i = 0; i< num; i++) { 5701 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5702 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5703 ref->mCnt--; 5704 ALOGV(" decremented refcount to %d", ref->mCnt); 5705 if (ref->mCnt == 0) { 5706 mAudioSessionRefs.removeAt(i); 5707 delete ref; 5708 purgeStaleEffects_l(); 5709 } 5710 return; 5711 } 5712 } 5713 ALOGW("session id %d not found for pid %d", audioSession, caller); 5714} 5715 5716void AudioFlinger::purgeStaleEffects_l() { 5717 5718 ALOGV("purging stale effects"); 5719 5720 Vector< sp<EffectChain> > chains; 5721 5722 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5723 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5724 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5725 sp<EffectChain> ec = t->mEffectChains[j]; 5726 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5727 chains.push(ec); 5728 } 5729 } 5730 } 5731 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5732 sp<RecordThread> t = mRecordThreads.valueAt(i); 5733 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5734 sp<EffectChain> ec = t->mEffectChains[j]; 5735 chains.push(ec); 5736 } 5737 } 5738 5739 for (size_t i = 0; i < chains.size(); i++) { 5740 sp<EffectChain> ec = chains[i]; 5741 int sessionid = ec->sessionId(); 5742 sp<ThreadBase> t = ec->mThread.promote(); 5743 if (t == 0) { 5744 continue; 5745 } 5746 size_t numsessionrefs = mAudioSessionRefs.size(); 5747 bool found = false; 5748 for (size_t k = 0; k < numsessionrefs; k++) { 5749 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5750 if (ref->mSessionid == sessionid) { 5751 ALOGV(" session %d still exists for %d with %d refs", 5752 sessionid, ref->mPid, ref->mCnt); 5753 found = true; 5754 break; 5755 } 5756 } 5757 if (!found) { 5758 // remove all effects from the chain 5759 while (ec->mEffects.size()) { 5760 sp<EffectModule> effect = ec->mEffects[0]; 5761 effect->unPin(); 5762 Mutex::Autolock _l (t->mLock); 5763 t->removeEffect_l(effect); 5764 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5765 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5766 if (handle != 0) { 5767 handle->mEffect.clear(); 5768 if (handle->mHasControl && handle->mEnabled) { 5769 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5770 } 5771 } 5772 } 5773 AudioSystem::unregisterEffect(effect->id()); 5774 } 5775 } 5776 } 5777 return; 5778} 5779 5780// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5781AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5782{ 5783 return mPlaybackThreads.valueFor(output).get(); 5784} 5785 5786// checkMixerThread_l() must be called with AudioFlinger::mLock held 5787AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5788{ 5789 PlaybackThread *thread = checkPlaybackThread_l(output); 5790 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5791} 5792 5793// checkRecordThread_l() must be called with AudioFlinger::mLock held 5794AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5795{ 5796 return mRecordThreads.valueFor(input).get(); 5797} 5798 5799uint32_t AudioFlinger::nextUniqueId() 5800{ 5801 return android_atomic_inc(&mNextUniqueId); 5802} 5803 5804AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5805{ 5806 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5807 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5808 AudioStreamOut *output = thread->getOutput(); 5809 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5810 return thread; 5811 } 5812 } 5813 return NULL; 5814} 5815 5816uint32_t AudioFlinger::primaryOutputDevice_l() const 5817{ 5818 PlaybackThread *thread = primaryPlaybackThread_l(); 5819 5820 if (thread == NULL) { 5821 return 0; 5822 } 5823 5824 return thread->device(); 5825} 5826 5827 5828// ---------------------------------------------------------------------------- 5829// Effect management 5830// ---------------------------------------------------------------------------- 5831 5832 5833status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5834{ 5835 Mutex::Autolock _l(mLock); 5836 return EffectQueryNumberEffects(numEffects); 5837} 5838 5839status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5840{ 5841 Mutex::Autolock _l(mLock); 5842 return EffectQueryEffect(index, descriptor); 5843} 5844 5845status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5846 effect_descriptor_t *descriptor) const 5847{ 5848 Mutex::Autolock _l(mLock); 5849 return EffectGetDescriptor(pUuid, descriptor); 5850} 5851 5852 5853sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5854 effect_descriptor_t *pDesc, 5855 const sp<IEffectClient>& effectClient, 5856 int32_t priority, 5857 audio_io_handle_t io, 5858 int sessionId, 5859 status_t *status, 5860 int *id, 5861 int *enabled) 5862{ 5863 status_t lStatus = NO_ERROR; 5864 sp<EffectHandle> handle; 5865 effect_descriptor_t desc; 5866 5867 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5868 pid, effectClient.get(), priority, sessionId, io); 5869 5870 if (pDesc == NULL) { 5871 lStatus = BAD_VALUE; 5872 goto Exit; 5873 } 5874 5875 // check audio settings permission for global effects 5876 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5877 lStatus = PERMISSION_DENIED; 5878 goto Exit; 5879 } 5880 5881 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5882 // that can only be created by audio policy manager (running in same process) 5883 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5884 lStatus = PERMISSION_DENIED; 5885 goto Exit; 5886 } 5887 5888 if (io == 0) { 5889 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5890 // output must be specified by AudioPolicyManager when using session 5891 // AUDIO_SESSION_OUTPUT_STAGE 5892 lStatus = BAD_VALUE; 5893 goto Exit; 5894 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5895 // if the output returned by getOutputForEffect() is removed before we lock the 5896 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5897 // and we will exit safely 5898 io = AudioSystem::getOutputForEffect(&desc); 5899 } 5900 } 5901 5902 { 5903 Mutex::Autolock _l(mLock); 5904 5905 5906 if (!EffectIsNullUuid(&pDesc->uuid)) { 5907 // if uuid is specified, request effect descriptor 5908 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5909 if (lStatus < 0) { 5910 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5911 goto Exit; 5912 } 5913 } else { 5914 // if uuid is not specified, look for an available implementation 5915 // of the required type in effect factory 5916 if (EffectIsNullUuid(&pDesc->type)) { 5917 ALOGW("createEffect() no effect type"); 5918 lStatus = BAD_VALUE; 5919 goto Exit; 5920 } 5921 uint32_t numEffects = 0; 5922 effect_descriptor_t d; 5923 d.flags = 0; // prevent compiler warning 5924 bool found = false; 5925 5926 lStatus = EffectQueryNumberEffects(&numEffects); 5927 if (lStatus < 0) { 5928 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5929 goto Exit; 5930 } 5931 for (uint32_t i = 0; i < numEffects; i++) { 5932 lStatus = EffectQueryEffect(i, &desc); 5933 if (lStatus < 0) { 5934 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5935 continue; 5936 } 5937 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5938 // If matching type found save effect descriptor. If the session is 5939 // 0 and the effect is not auxiliary, continue enumeration in case 5940 // an auxiliary version of this effect type is available 5941 found = true; 5942 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5943 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5944 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5945 break; 5946 } 5947 } 5948 } 5949 if (!found) { 5950 lStatus = BAD_VALUE; 5951 ALOGW("createEffect() effect not found"); 5952 goto Exit; 5953 } 5954 // For same effect type, chose auxiliary version over insert version if 5955 // connect to output mix (Compliance to OpenSL ES) 5956 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5957 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5958 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5959 } 5960 } 5961 5962 // Do not allow auxiliary effects on a session different from 0 (output mix) 5963 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5964 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5965 lStatus = INVALID_OPERATION; 5966 goto Exit; 5967 } 5968 5969 // check recording permission for visualizer 5970 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5971 !recordingAllowed()) { 5972 lStatus = PERMISSION_DENIED; 5973 goto Exit; 5974 } 5975 5976 // return effect descriptor 5977 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5978 5979 // If output is not specified try to find a matching audio session ID in one of the 5980 // output threads. 5981 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5982 // because of code checking output when entering the function. 5983 // Note: io is never 0 when creating an effect on an input 5984 if (io == 0) { 5985 // look for the thread where the specified audio session is present 5986 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5987 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5988 io = mPlaybackThreads.keyAt(i); 5989 break; 5990 } 5991 } 5992 if (io == 0) { 5993 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5994 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5995 io = mRecordThreads.keyAt(i); 5996 break; 5997 } 5998 } 5999 } 6000 // If no output thread contains the requested session ID, default to 6001 // first output. The effect chain will be moved to the correct output 6002 // thread when a track with the same session ID is created 6003 if (io == 0 && mPlaybackThreads.size()) { 6004 io = mPlaybackThreads.keyAt(0); 6005 } 6006 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6007 } 6008 ThreadBase *thread = checkRecordThread_l(io); 6009 if (thread == NULL) { 6010 thread = checkPlaybackThread_l(io); 6011 if (thread == NULL) { 6012 ALOGE("createEffect() unknown output thread"); 6013 lStatus = BAD_VALUE; 6014 goto Exit; 6015 } 6016 } 6017 6018 sp<Client> client = registerPid_l(pid); 6019 6020 // create effect on selected output thread 6021 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6022 &desc, enabled, &lStatus); 6023 if (handle != 0 && id != NULL) { 6024 *id = handle->id(); 6025 } 6026 } 6027 6028Exit: 6029 if(status) { 6030 *status = lStatus; 6031 } 6032 return handle; 6033} 6034 6035status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6036 audio_io_handle_t dstOutput) 6037{ 6038 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6039 sessionId, srcOutput, dstOutput); 6040 Mutex::Autolock _l(mLock); 6041 if (srcOutput == dstOutput) { 6042 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6043 return NO_ERROR; 6044 } 6045 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6046 if (srcThread == NULL) { 6047 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6048 return BAD_VALUE; 6049 } 6050 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6051 if (dstThread == NULL) { 6052 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6053 return BAD_VALUE; 6054 } 6055 6056 Mutex::Autolock _dl(dstThread->mLock); 6057 Mutex::Autolock _sl(srcThread->mLock); 6058 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6059 6060 return NO_ERROR; 6061} 6062 6063// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6064status_t AudioFlinger::moveEffectChain_l(int sessionId, 6065 AudioFlinger::PlaybackThread *srcThread, 6066 AudioFlinger::PlaybackThread *dstThread, 6067 bool reRegister) 6068{ 6069 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6070 sessionId, srcThread, dstThread); 6071 6072 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6073 if (chain == 0) { 6074 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6075 sessionId, srcThread); 6076 return INVALID_OPERATION; 6077 } 6078 6079 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6080 // so that a new chain is created with correct parameters when first effect is added. This is 6081 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6082 // removed. 6083 srcThread->removeEffectChain_l(chain); 6084 6085 // transfer all effects one by one so that new effect chain is created on new thread with 6086 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6087 audio_io_handle_t dstOutput = dstThread->id(); 6088 sp<EffectChain> dstChain; 6089 uint32_t strategy = 0; // prevent compiler warning 6090 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6091 while (effect != 0) { 6092 srcThread->removeEffect_l(effect); 6093 dstThread->addEffect_l(effect); 6094 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6095 if (effect->state() == EffectModule::ACTIVE || 6096 effect->state() == EffectModule::STOPPING) { 6097 effect->start(); 6098 } 6099 // if the move request is not received from audio policy manager, the effect must be 6100 // re-registered with the new strategy and output 6101 if (dstChain == 0) { 6102 dstChain = effect->chain().promote(); 6103 if (dstChain == 0) { 6104 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6105 srcThread->addEffect_l(effect); 6106 return NO_INIT; 6107 } 6108 strategy = dstChain->strategy(); 6109 } 6110 if (reRegister) { 6111 AudioSystem::unregisterEffect(effect->id()); 6112 AudioSystem::registerEffect(&effect->desc(), 6113 dstOutput, 6114 strategy, 6115 sessionId, 6116 effect->id()); 6117 } 6118 effect = chain->getEffectFromId_l(0); 6119 } 6120 6121 return NO_ERROR; 6122} 6123 6124 6125// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6126sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6127 const sp<AudioFlinger::Client>& client, 6128 const sp<IEffectClient>& effectClient, 6129 int32_t priority, 6130 int sessionId, 6131 effect_descriptor_t *desc, 6132 int *enabled, 6133 status_t *status 6134 ) 6135{ 6136 sp<EffectModule> effect; 6137 sp<EffectHandle> handle; 6138 status_t lStatus; 6139 sp<EffectChain> chain; 6140 bool chainCreated = false; 6141 bool effectCreated = false; 6142 bool effectRegistered = false; 6143 6144 lStatus = initCheck(); 6145 if (lStatus != NO_ERROR) { 6146 ALOGW("createEffect_l() Audio driver not initialized."); 6147 goto Exit; 6148 } 6149 6150 // Do not allow effects with session ID 0 on direct output or duplicating threads 6151 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6152 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6153 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6154 desc->name, sessionId); 6155 lStatus = BAD_VALUE; 6156 goto Exit; 6157 } 6158 // Only Pre processor effects are allowed on input threads and only on input threads 6159 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6160 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6161 desc->name, desc->flags, mType); 6162 lStatus = BAD_VALUE; 6163 goto Exit; 6164 } 6165 6166 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6167 6168 { // scope for mLock 6169 Mutex::Autolock _l(mLock); 6170 6171 // check for existing effect chain with the requested audio session 6172 chain = getEffectChain_l(sessionId); 6173 if (chain == 0) { 6174 // create a new chain for this session 6175 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6176 chain = new EffectChain(this, sessionId); 6177 addEffectChain_l(chain); 6178 chain->setStrategy(getStrategyForSession_l(sessionId)); 6179 chainCreated = true; 6180 } else { 6181 effect = chain->getEffectFromDesc_l(desc); 6182 } 6183 6184 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6185 6186 if (effect == 0) { 6187 int id = mAudioFlinger->nextUniqueId(); 6188 // Check CPU and memory usage 6189 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6190 if (lStatus != NO_ERROR) { 6191 goto Exit; 6192 } 6193 effectRegistered = true; 6194 // create a new effect module if none present in the chain 6195 effect = new EffectModule(this, chain, desc, id, sessionId); 6196 lStatus = effect->status(); 6197 if (lStatus != NO_ERROR) { 6198 goto Exit; 6199 } 6200 lStatus = chain->addEffect_l(effect); 6201 if (lStatus != NO_ERROR) { 6202 goto Exit; 6203 } 6204 effectCreated = true; 6205 6206 effect->setDevice(mDevice); 6207 effect->setMode(mAudioFlinger->getMode()); 6208 } 6209 // create effect handle and connect it to effect module 6210 handle = new EffectHandle(effect, client, effectClient, priority); 6211 lStatus = effect->addHandle(handle); 6212 if (enabled != NULL) { 6213 *enabled = (int)effect->isEnabled(); 6214 } 6215 } 6216 6217Exit: 6218 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6219 Mutex::Autolock _l(mLock); 6220 if (effectCreated) { 6221 chain->removeEffect_l(effect); 6222 } 6223 if (effectRegistered) { 6224 AudioSystem::unregisterEffect(effect->id()); 6225 } 6226 if (chainCreated) { 6227 removeEffectChain_l(chain); 6228 } 6229 handle.clear(); 6230 } 6231 6232 if(status) { 6233 *status = lStatus; 6234 } 6235 return handle; 6236} 6237 6238sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6239{ 6240 sp<EffectChain> chain = getEffectChain_l(sessionId); 6241 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6242} 6243 6244// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6245// PlaybackThread::mLock held 6246status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6247{ 6248 // check for existing effect chain with the requested audio session 6249 int sessionId = effect->sessionId(); 6250 sp<EffectChain> chain = getEffectChain_l(sessionId); 6251 bool chainCreated = false; 6252 6253 if (chain == 0) { 6254 // create a new chain for this session 6255 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6256 chain = new EffectChain(this, sessionId); 6257 addEffectChain_l(chain); 6258 chain->setStrategy(getStrategyForSession_l(sessionId)); 6259 chainCreated = true; 6260 } 6261 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6262 6263 if (chain->getEffectFromId_l(effect->id()) != 0) { 6264 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6265 this, effect->desc().name, chain.get()); 6266 return BAD_VALUE; 6267 } 6268 6269 status_t status = chain->addEffect_l(effect); 6270 if (status != NO_ERROR) { 6271 if (chainCreated) { 6272 removeEffectChain_l(chain); 6273 } 6274 return status; 6275 } 6276 6277 effect->setDevice(mDevice); 6278 effect->setMode(mAudioFlinger->getMode()); 6279 return NO_ERROR; 6280} 6281 6282void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6283 6284 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6285 effect_descriptor_t desc = effect->desc(); 6286 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6287 detachAuxEffect_l(effect->id()); 6288 } 6289 6290 sp<EffectChain> chain = effect->chain().promote(); 6291 if (chain != 0) { 6292 // remove effect chain if removing last effect 6293 if (chain->removeEffect_l(effect) == 0) { 6294 removeEffectChain_l(chain); 6295 } 6296 } else { 6297 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6298 } 6299} 6300 6301void AudioFlinger::ThreadBase::lockEffectChains_l( 6302 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6303{ 6304 effectChains = mEffectChains; 6305 for (size_t i = 0; i < mEffectChains.size(); i++) { 6306 mEffectChains[i]->lock(); 6307 } 6308} 6309 6310void AudioFlinger::ThreadBase::unlockEffectChains( 6311 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6312{ 6313 for (size_t i = 0; i < effectChains.size(); i++) { 6314 effectChains[i]->unlock(); 6315 } 6316} 6317 6318sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6319{ 6320 Mutex::Autolock _l(mLock); 6321 return getEffectChain_l(sessionId); 6322} 6323 6324sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6325{ 6326 size_t size = mEffectChains.size(); 6327 for (size_t i = 0; i < size; i++) { 6328 if (mEffectChains[i]->sessionId() == sessionId) { 6329 return mEffectChains[i]; 6330 } 6331 } 6332 return 0; 6333} 6334 6335void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6336{ 6337 Mutex::Autolock _l(mLock); 6338 size_t size = mEffectChains.size(); 6339 for (size_t i = 0; i < size; i++) { 6340 mEffectChains[i]->setMode_l(mode); 6341 } 6342} 6343 6344void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6345 const wp<EffectHandle>& handle, 6346 bool unpinIfLast) { 6347 6348 Mutex::Autolock _l(mLock); 6349 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6350 // delete the effect module if removing last handle on it 6351 if (effect->removeHandle(handle) == 0) { 6352 if (!effect->isPinned() || unpinIfLast) { 6353 removeEffect_l(effect); 6354 AudioSystem::unregisterEffect(effect->id()); 6355 } 6356 } 6357} 6358 6359status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6360{ 6361 int session = chain->sessionId(); 6362 int16_t *buffer = mMixBuffer; 6363 bool ownsBuffer = false; 6364 6365 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6366 if (session > 0) { 6367 // Only one effect chain can be present in direct output thread and it uses 6368 // the mix buffer as input 6369 if (mType != DIRECT) { 6370 size_t numSamples = mFrameCount * mChannelCount; 6371 buffer = new int16_t[numSamples]; 6372 memset(buffer, 0, numSamples * sizeof(int16_t)); 6373 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6374 ownsBuffer = true; 6375 } 6376 6377 // Attach all tracks with same session ID to this chain. 6378 for (size_t i = 0; i < mTracks.size(); ++i) { 6379 sp<Track> track = mTracks[i]; 6380 if (session == track->sessionId()) { 6381 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6382 track->setMainBuffer(buffer); 6383 chain->incTrackCnt(); 6384 } 6385 } 6386 6387 // indicate all active tracks in the chain 6388 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6389 sp<Track> track = mActiveTracks[i].promote(); 6390 if (track == 0) continue; 6391 if (session == track->sessionId()) { 6392 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6393 chain->incActiveTrackCnt(); 6394 } 6395 } 6396 } 6397 6398 chain->setInBuffer(buffer, ownsBuffer); 6399 chain->setOutBuffer(mMixBuffer); 6400 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6401 // chains list in order to be processed last as it contains output stage effects 6402 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6403 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6404 // after track specific effects and before output stage 6405 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6406 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6407 // Effect chain for other sessions are inserted at beginning of effect 6408 // chains list to be processed before output mix effects. Relative order between other 6409 // sessions is not important 6410 size_t size = mEffectChains.size(); 6411 size_t i = 0; 6412 for (i = 0; i < size; i++) { 6413 if (mEffectChains[i]->sessionId() < session) break; 6414 } 6415 mEffectChains.insertAt(chain, i); 6416 checkSuspendOnAddEffectChain_l(chain); 6417 6418 return NO_ERROR; 6419} 6420 6421size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6422{ 6423 int session = chain->sessionId(); 6424 6425 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6426 6427 for (size_t i = 0; i < mEffectChains.size(); i++) { 6428 if (chain == mEffectChains[i]) { 6429 mEffectChains.removeAt(i); 6430 // detach all active tracks from the chain 6431 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6432 sp<Track> track = mActiveTracks[i].promote(); 6433 if (track == 0) continue; 6434 if (session == track->sessionId()) { 6435 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6436 chain.get(), session); 6437 chain->decActiveTrackCnt(); 6438 } 6439 } 6440 6441 // detach all tracks with same session ID from this chain 6442 for (size_t i = 0; i < mTracks.size(); ++i) { 6443 sp<Track> track = mTracks[i]; 6444 if (session == track->sessionId()) { 6445 track->setMainBuffer(mMixBuffer); 6446 chain->decTrackCnt(); 6447 } 6448 } 6449 break; 6450 } 6451 } 6452 return mEffectChains.size(); 6453} 6454 6455status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6456 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6457{ 6458 Mutex::Autolock _l(mLock); 6459 return attachAuxEffect_l(track, EffectId); 6460} 6461 6462status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6463 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6464{ 6465 status_t status = NO_ERROR; 6466 6467 if (EffectId == 0) { 6468 track->setAuxBuffer(0, NULL); 6469 } else { 6470 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6471 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6472 if (effect != 0) { 6473 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6474 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6475 } else { 6476 status = INVALID_OPERATION; 6477 } 6478 } else { 6479 status = BAD_VALUE; 6480 } 6481 } 6482 return status; 6483} 6484 6485void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6486{ 6487 for (size_t i = 0; i < mTracks.size(); ++i) { 6488 sp<Track> track = mTracks[i]; 6489 if (track->auxEffectId() == effectId) { 6490 attachAuxEffect_l(track, 0); 6491 } 6492 } 6493} 6494 6495status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6496{ 6497 // only one chain per input thread 6498 if (mEffectChains.size() != 0) { 6499 return INVALID_OPERATION; 6500 } 6501 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6502 6503 chain->setInBuffer(NULL); 6504 chain->setOutBuffer(NULL); 6505 6506 checkSuspendOnAddEffectChain_l(chain); 6507 6508 mEffectChains.add(chain); 6509 6510 return NO_ERROR; 6511} 6512 6513size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6514{ 6515 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6516 ALOGW_IF(mEffectChains.size() != 1, 6517 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6518 chain.get(), mEffectChains.size(), this); 6519 if (mEffectChains.size() == 1) { 6520 mEffectChains.removeAt(0); 6521 } 6522 return 0; 6523} 6524 6525// ---------------------------------------------------------------------------- 6526// EffectModule implementation 6527// ---------------------------------------------------------------------------- 6528 6529#undef LOG_TAG 6530#define LOG_TAG "AudioFlinger::EffectModule" 6531 6532AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6533 const wp<AudioFlinger::EffectChain>& chain, 6534 effect_descriptor_t *desc, 6535 int id, 6536 int sessionId) 6537 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6538 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6539{ 6540 ALOGV("Constructor %p", this); 6541 int lStatus; 6542 if (thread == NULL) { 6543 return; 6544 } 6545 6546 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6547 6548 // create effect engine from effect factory 6549 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6550 6551 if (mStatus != NO_ERROR) { 6552 return; 6553 } 6554 lStatus = init(); 6555 if (lStatus < 0) { 6556 mStatus = lStatus; 6557 goto Error; 6558 } 6559 6560 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6561 mPinned = true; 6562 } 6563 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6564 return; 6565Error: 6566 EffectRelease(mEffectInterface); 6567 mEffectInterface = NULL; 6568 ALOGV("Constructor Error %d", mStatus); 6569} 6570 6571AudioFlinger::EffectModule::~EffectModule() 6572{ 6573 ALOGV("Destructor %p", this); 6574 if (mEffectInterface != NULL) { 6575 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6576 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6577 sp<ThreadBase> thread = mThread.promote(); 6578 if (thread != 0) { 6579 audio_stream_t *stream = thread->stream(); 6580 if (stream != NULL) { 6581 stream->remove_audio_effect(stream, mEffectInterface); 6582 } 6583 } 6584 } 6585 // release effect engine 6586 EffectRelease(mEffectInterface); 6587 } 6588} 6589 6590status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6591{ 6592 status_t status; 6593 6594 Mutex::Autolock _l(mLock); 6595 int priority = handle->priority(); 6596 size_t size = mHandles.size(); 6597 sp<EffectHandle> h; 6598 size_t i; 6599 for (i = 0; i < size; i++) { 6600 h = mHandles[i].promote(); 6601 if (h == 0) continue; 6602 if (h->priority() <= priority) break; 6603 } 6604 // if inserted in first place, move effect control from previous owner to this handle 6605 if (i == 0) { 6606 bool enabled = false; 6607 if (h != 0) { 6608 enabled = h->enabled(); 6609 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6610 } 6611 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6612 status = NO_ERROR; 6613 } else { 6614 status = ALREADY_EXISTS; 6615 } 6616 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6617 mHandles.insertAt(handle, i); 6618 return status; 6619} 6620 6621size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6622{ 6623 Mutex::Autolock _l(mLock); 6624 size_t size = mHandles.size(); 6625 size_t i; 6626 for (i = 0; i < size; i++) { 6627 if (mHandles[i] == handle) break; 6628 } 6629 if (i == size) { 6630 return size; 6631 } 6632 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6633 6634 bool enabled = false; 6635 EffectHandle *hdl = handle.unsafe_get(); 6636 if (hdl != NULL) { 6637 ALOGV("removeHandle() unsafe_get OK"); 6638 enabled = hdl->enabled(); 6639 } 6640 mHandles.removeAt(i); 6641 size = mHandles.size(); 6642 // if removed from first place, move effect control from this handle to next in line 6643 if (i == 0 && size != 0) { 6644 sp<EffectHandle> h = mHandles[0].promote(); 6645 if (h != 0) { 6646 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6647 } 6648 } 6649 6650 // Prevent calls to process() and other functions on effect interface from now on. 6651 // The effect engine will be released by the destructor when the last strong reference on 6652 // this object is released which can happen after next process is called. 6653 if (size == 0 && !mPinned) { 6654 mState = DESTROYED; 6655 } 6656 6657 return size; 6658} 6659 6660sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6661{ 6662 Mutex::Autolock _l(mLock); 6663 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6664} 6665 6666void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6667{ 6668 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6669 // keep a strong reference on this EffectModule to avoid calling the 6670 // destructor before we exit 6671 sp<EffectModule> keep(this); 6672 { 6673 sp<ThreadBase> thread = mThread.promote(); 6674 if (thread != 0) { 6675 thread->disconnectEffect(keep, handle, unpinIfLast); 6676 } 6677 } 6678} 6679 6680void AudioFlinger::EffectModule::updateState() { 6681 Mutex::Autolock _l(mLock); 6682 6683 switch (mState) { 6684 case RESTART: 6685 reset_l(); 6686 // FALL THROUGH 6687 6688 case STARTING: 6689 // clear auxiliary effect input buffer for next accumulation 6690 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6691 memset(mConfig.inputCfg.buffer.raw, 6692 0, 6693 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6694 } 6695 start_l(); 6696 mState = ACTIVE; 6697 break; 6698 case STOPPING: 6699 stop_l(); 6700 mDisableWaitCnt = mMaxDisableWaitCnt; 6701 mState = STOPPED; 6702 break; 6703 case STOPPED: 6704 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6705 // turn off sequence. 6706 if (--mDisableWaitCnt == 0) { 6707 reset_l(); 6708 mState = IDLE; 6709 } 6710 break; 6711 default: //IDLE , ACTIVE, DESTROYED 6712 break; 6713 } 6714} 6715 6716void AudioFlinger::EffectModule::process() 6717{ 6718 Mutex::Autolock _l(mLock); 6719 6720 if (mState == DESTROYED || mEffectInterface == NULL || 6721 mConfig.inputCfg.buffer.raw == NULL || 6722 mConfig.outputCfg.buffer.raw == NULL) { 6723 return; 6724 } 6725 6726 if (isProcessEnabled()) { 6727 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6728 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6729 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6730 mConfig.inputCfg.buffer.s32, 6731 mConfig.inputCfg.buffer.frameCount/2); 6732 } 6733 6734 // do the actual processing in the effect engine 6735 int ret = (*mEffectInterface)->process(mEffectInterface, 6736 &mConfig.inputCfg.buffer, 6737 &mConfig.outputCfg.buffer); 6738 6739 // force transition to IDLE state when engine is ready 6740 if (mState == STOPPED && ret == -ENODATA) { 6741 mDisableWaitCnt = 1; 6742 } 6743 6744 // clear auxiliary effect input buffer for next accumulation 6745 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6746 memset(mConfig.inputCfg.buffer.raw, 0, 6747 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6748 } 6749 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6750 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6751 // If an insert effect is idle and input buffer is different from output buffer, 6752 // accumulate input onto output 6753 sp<EffectChain> chain = mChain.promote(); 6754 if (chain != 0 && chain->activeTrackCnt() != 0) { 6755 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6756 int16_t *in = mConfig.inputCfg.buffer.s16; 6757 int16_t *out = mConfig.outputCfg.buffer.s16; 6758 for (size_t i = 0; i < frameCnt; i++) { 6759 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6760 } 6761 } 6762 } 6763} 6764 6765void AudioFlinger::EffectModule::reset_l() 6766{ 6767 if (mEffectInterface == NULL) { 6768 return; 6769 } 6770 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6771} 6772 6773status_t AudioFlinger::EffectModule::configure() 6774{ 6775 uint32_t channels; 6776 if (mEffectInterface == NULL) { 6777 return NO_INIT; 6778 } 6779 6780 sp<ThreadBase> thread = mThread.promote(); 6781 if (thread == 0) { 6782 return DEAD_OBJECT; 6783 } 6784 6785 // TODO: handle configuration of effects replacing track process 6786 if (thread->channelCount() == 1) { 6787 channels = AUDIO_CHANNEL_OUT_MONO; 6788 } else { 6789 channels = AUDIO_CHANNEL_OUT_STEREO; 6790 } 6791 6792 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6793 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6794 } else { 6795 mConfig.inputCfg.channels = channels; 6796 } 6797 mConfig.outputCfg.channels = channels; 6798 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6799 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6800 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6801 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6802 mConfig.inputCfg.bufferProvider.cookie = NULL; 6803 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6804 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6805 mConfig.outputCfg.bufferProvider.cookie = NULL; 6806 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6807 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6808 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6809 // Insert effect: 6810 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6811 // always overwrites output buffer: input buffer == output buffer 6812 // - in other sessions: 6813 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6814 // other effect: overwrites output buffer: input buffer == output buffer 6815 // Auxiliary effect: 6816 // accumulates in output buffer: input buffer != output buffer 6817 // Therefore: accumulate <=> input buffer != output buffer 6818 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6819 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6820 } else { 6821 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6822 } 6823 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6824 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6825 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6826 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6827 6828 ALOGV("configure() %p thread %p buffer %p framecount %d", 6829 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6830 6831 status_t cmdStatus; 6832 uint32_t size = sizeof(int); 6833 status_t status = (*mEffectInterface)->command(mEffectInterface, 6834 EFFECT_CMD_SET_CONFIG, 6835 sizeof(effect_config_t), 6836 &mConfig, 6837 &size, 6838 &cmdStatus); 6839 if (status == 0) { 6840 status = cmdStatus; 6841 } 6842 6843 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6844 (1000 * mConfig.outputCfg.buffer.frameCount); 6845 6846 return status; 6847} 6848 6849status_t AudioFlinger::EffectModule::init() 6850{ 6851 Mutex::Autolock _l(mLock); 6852 if (mEffectInterface == NULL) { 6853 return NO_INIT; 6854 } 6855 status_t cmdStatus; 6856 uint32_t size = sizeof(status_t); 6857 status_t status = (*mEffectInterface)->command(mEffectInterface, 6858 EFFECT_CMD_INIT, 6859 0, 6860 NULL, 6861 &size, 6862 &cmdStatus); 6863 if (status == 0) { 6864 status = cmdStatus; 6865 } 6866 return status; 6867} 6868 6869status_t AudioFlinger::EffectModule::start() 6870{ 6871 Mutex::Autolock _l(mLock); 6872 return start_l(); 6873} 6874 6875status_t AudioFlinger::EffectModule::start_l() 6876{ 6877 if (mEffectInterface == NULL) { 6878 return NO_INIT; 6879 } 6880 status_t cmdStatus; 6881 uint32_t size = sizeof(status_t); 6882 status_t status = (*mEffectInterface)->command(mEffectInterface, 6883 EFFECT_CMD_ENABLE, 6884 0, 6885 NULL, 6886 &size, 6887 &cmdStatus); 6888 if (status == 0) { 6889 status = cmdStatus; 6890 } 6891 if (status == 0 && 6892 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6893 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6894 sp<ThreadBase> thread = mThread.promote(); 6895 if (thread != 0) { 6896 audio_stream_t *stream = thread->stream(); 6897 if (stream != NULL) { 6898 stream->add_audio_effect(stream, mEffectInterface); 6899 } 6900 } 6901 } 6902 return status; 6903} 6904 6905status_t AudioFlinger::EffectModule::stop() 6906{ 6907 Mutex::Autolock _l(mLock); 6908 return stop_l(); 6909} 6910 6911status_t AudioFlinger::EffectModule::stop_l() 6912{ 6913 if (mEffectInterface == NULL) { 6914 return NO_INIT; 6915 } 6916 status_t cmdStatus; 6917 uint32_t size = sizeof(status_t); 6918 status_t status = (*mEffectInterface)->command(mEffectInterface, 6919 EFFECT_CMD_DISABLE, 6920 0, 6921 NULL, 6922 &size, 6923 &cmdStatus); 6924 if (status == 0) { 6925 status = cmdStatus; 6926 } 6927 if (status == 0 && 6928 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6929 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6930 sp<ThreadBase> thread = mThread.promote(); 6931 if (thread != 0) { 6932 audio_stream_t *stream = thread->stream(); 6933 if (stream != NULL) { 6934 stream->remove_audio_effect(stream, mEffectInterface); 6935 } 6936 } 6937 } 6938 return status; 6939} 6940 6941status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6942 uint32_t cmdSize, 6943 void *pCmdData, 6944 uint32_t *replySize, 6945 void *pReplyData) 6946{ 6947 Mutex::Autolock _l(mLock); 6948// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6949 6950 if (mState == DESTROYED || mEffectInterface == NULL) { 6951 return NO_INIT; 6952 } 6953 status_t status = (*mEffectInterface)->command(mEffectInterface, 6954 cmdCode, 6955 cmdSize, 6956 pCmdData, 6957 replySize, 6958 pReplyData); 6959 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6960 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6961 for (size_t i = 1; i < mHandles.size(); i++) { 6962 sp<EffectHandle> h = mHandles[i].promote(); 6963 if (h != 0) { 6964 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6965 } 6966 } 6967 } 6968 return status; 6969} 6970 6971status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6972{ 6973 6974 Mutex::Autolock _l(mLock); 6975 ALOGV("setEnabled %p enabled %d", this, enabled); 6976 6977 if (enabled != isEnabled()) { 6978 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6979 if (enabled && status != NO_ERROR) { 6980 return status; 6981 } 6982 6983 switch (mState) { 6984 // going from disabled to enabled 6985 case IDLE: 6986 mState = STARTING; 6987 break; 6988 case STOPPED: 6989 mState = RESTART; 6990 break; 6991 case STOPPING: 6992 mState = ACTIVE; 6993 break; 6994 6995 // going from enabled to disabled 6996 case RESTART: 6997 mState = STOPPED; 6998 break; 6999 case STARTING: 7000 mState = IDLE; 7001 break; 7002 case ACTIVE: 7003 mState = STOPPING; 7004 break; 7005 case DESTROYED: 7006 return NO_ERROR; // simply ignore as we are being destroyed 7007 } 7008 for (size_t i = 1; i < mHandles.size(); i++) { 7009 sp<EffectHandle> h = mHandles[i].promote(); 7010 if (h != 0) { 7011 h->setEnabled(enabled); 7012 } 7013 } 7014 } 7015 return NO_ERROR; 7016} 7017 7018bool AudioFlinger::EffectModule::isEnabled() const 7019{ 7020 switch (mState) { 7021 case RESTART: 7022 case STARTING: 7023 case ACTIVE: 7024 return true; 7025 case IDLE: 7026 case STOPPING: 7027 case STOPPED: 7028 case DESTROYED: 7029 default: 7030 return false; 7031 } 7032} 7033 7034bool AudioFlinger::EffectModule::isProcessEnabled() const 7035{ 7036 switch (mState) { 7037 case RESTART: 7038 case ACTIVE: 7039 case STOPPING: 7040 case STOPPED: 7041 return true; 7042 case IDLE: 7043 case STARTING: 7044 case DESTROYED: 7045 default: 7046 return false; 7047 } 7048} 7049 7050status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7051{ 7052 Mutex::Autolock _l(mLock); 7053 status_t status = NO_ERROR; 7054 7055 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7056 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7057 if (isProcessEnabled() && 7058 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7059 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7060 status_t cmdStatus; 7061 uint32_t volume[2]; 7062 uint32_t *pVolume = NULL; 7063 uint32_t size = sizeof(volume); 7064 volume[0] = *left; 7065 volume[1] = *right; 7066 if (controller) { 7067 pVolume = volume; 7068 } 7069 status = (*mEffectInterface)->command(mEffectInterface, 7070 EFFECT_CMD_SET_VOLUME, 7071 size, 7072 volume, 7073 &size, 7074 pVolume); 7075 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7076 *left = volume[0]; 7077 *right = volume[1]; 7078 } 7079 } 7080 return status; 7081} 7082 7083status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7084{ 7085 Mutex::Autolock _l(mLock); 7086 status_t status = NO_ERROR; 7087 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7088 // audio pre processing modules on RecordThread can receive both output and 7089 // input device indication in the same call 7090 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7091 if (dev) { 7092 status_t cmdStatus; 7093 uint32_t size = sizeof(status_t); 7094 7095 status = (*mEffectInterface)->command(mEffectInterface, 7096 EFFECT_CMD_SET_DEVICE, 7097 sizeof(uint32_t), 7098 &dev, 7099 &size, 7100 &cmdStatus); 7101 if (status == NO_ERROR) { 7102 status = cmdStatus; 7103 } 7104 } 7105 dev = device & AUDIO_DEVICE_IN_ALL; 7106 if (dev) { 7107 status_t cmdStatus; 7108 uint32_t size = sizeof(status_t); 7109 7110 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7111 EFFECT_CMD_SET_INPUT_DEVICE, 7112 sizeof(uint32_t), 7113 &dev, 7114 &size, 7115 &cmdStatus); 7116 if (status2 == NO_ERROR) { 7117 status2 = cmdStatus; 7118 } 7119 if (status == NO_ERROR) { 7120 status = status2; 7121 } 7122 } 7123 } 7124 return status; 7125} 7126 7127status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7128{ 7129 Mutex::Autolock _l(mLock); 7130 status_t status = NO_ERROR; 7131 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7132 status_t cmdStatus; 7133 uint32_t size = sizeof(status_t); 7134 status = (*mEffectInterface)->command(mEffectInterface, 7135 EFFECT_CMD_SET_AUDIO_MODE, 7136 sizeof(audio_mode_t), 7137 &mode, 7138 &size, 7139 &cmdStatus); 7140 if (status == NO_ERROR) { 7141 status = cmdStatus; 7142 } 7143 } 7144 return status; 7145} 7146 7147void AudioFlinger::EffectModule::setSuspended(bool suspended) 7148{ 7149 Mutex::Autolock _l(mLock); 7150 mSuspended = suspended; 7151} 7152 7153bool AudioFlinger::EffectModule::suspended() const 7154{ 7155 Mutex::Autolock _l(mLock); 7156 return mSuspended; 7157} 7158 7159status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7160{ 7161 const size_t SIZE = 256; 7162 char buffer[SIZE]; 7163 String8 result; 7164 7165 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7166 result.append(buffer); 7167 7168 bool locked = tryLock(mLock); 7169 // failed to lock - AudioFlinger is probably deadlocked 7170 if (!locked) { 7171 result.append("\t\tCould not lock Fx mutex:\n"); 7172 } 7173 7174 result.append("\t\tSession Status State Engine:\n"); 7175 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7176 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7177 result.append(buffer); 7178 7179 result.append("\t\tDescriptor:\n"); 7180 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7181 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7182 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7183 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7184 result.append(buffer); 7185 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7186 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7187 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7188 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7189 result.append(buffer); 7190 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7191 mDescriptor.apiVersion, 7192 mDescriptor.flags); 7193 result.append(buffer); 7194 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7195 mDescriptor.name); 7196 result.append(buffer); 7197 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7198 mDescriptor.implementor); 7199 result.append(buffer); 7200 7201 result.append("\t\t- Input configuration:\n"); 7202 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7203 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7204 (uint32_t)mConfig.inputCfg.buffer.raw, 7205 mConfig.inputCfg.buffer.frameCount, 7206 mConfig.inputCfg.samplingRate, 7207 mConfig.inputCfg.channels, 7208 mConfig.inputCfg.format); 7209 result.append(buffer); 7210 7211 result.append("\t\t- Output configuration:\n"); 7212 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7213 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7214 (uint32_t)mConfig.outputCfg.buffer.raw, 7215 mConfig.outputCfg.buffer.frameCount, 7216 mConfig.outputCfg.samplingRate, 7217 mConfig.outputCfg.channels, 7218 mConfig.outputCfg.format); 7219 result.append(buffer); 7220 7221 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7222 result.append(buffer); 7223 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7224 for (size_t i = 0; i < mHandles.size(); ++i) { 7225 sp<EffectHandle> handle = mHandles[i].promote(); 7226 if (handle != 0) { 7227 handle->dump(buffer, SIZE); 7228 result.append(buffer); 7229 } 7230 } 7231 7232 result.append("\n"); 7233 7234 write(fd, result.string(), result.length()); 7235 7236 if (locked) { 7237 mLock.unlock(); 7238 } 7239 7240 return NO_ERROR; 7241} 7242 7243// ---------------------------------------------------------------------------- 7244// EffectHandle implementation 7245// ---------------------------------------------------------------------------- 7246 7247#undef LOG_TAG 7248#define LOG_TAG "AudioFlinger::EffectHandle" 7249 7250AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7251 const sp<AudioFlinger::Client>& client, 7252 const sp<IEffectClient>& effectClient, 7253 int32_t priority) 7254 : BnEffect(), 7255 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7256 mPriority(priority), mHasControl(false), mEnabled(false) 7257{ 7258 ALOGV("constructor %p", this); 7259 7260 if (client == 0) { 7261 return; 7262 } 7263 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7264 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7265 if (mCblkMemory != 0) { 7266 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7267 7268 if (mCblk != NULL) { 7269 new(mCblk) effect_param_cblk_t(); 7270 mBuffer = (uint8_t *)mCblk + bufOffset; 7271 } 7272 } else { 7273 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7274 return; 7275 } 7276} 7277 7278AudioFlinger::EffectHandle::~EffectHandle() 7279{ 7280 ALOGV("Destructor %p", this); 7281 disconnect(false); 7282 ALOGV("Destructor DONE %p", this); 7283} 7284 7285status_t AudioFlinger::EffectHandle::enable() 7286{ 7287 ALOGV("enable %p", this); 7288 if (!mHasControl) return INVALID_OPERATION; 7289 if (mEffect == 0) return DEAD_OBJECT; 7290 7291 if (mEnabled) { 7292 return NO_ERROR; 7293 } 7294 7295 mEnabled = true; 7296 7297 sp<ThreadBase> thread = mEffect->thread().promote(); 7298 if (thread != 0) { 7299 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7300 } 7301 7302 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7303 if (mEffect->suspended()) { 7304 return NO_ERROR; 7305 } 7306 7307 status_t status = mEffect->setEnabled(true); 7308 if (status != NO_ERROR) { 7309 if (thread != 0) { 7310 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7311 } 7312 mEnabled = false; 7313 } 7314 return status; 7315} 7316 7317status_t AudioFlinger::EffectHandle::disable() 7318{ 7319 ALOGV("disable %p", this); 7320 if (!mHasControl) return INVALID_OPERATION; 7321 if (mEffect == 0) return DEAD_OBJECT; 7322 7323 if (!mEnabled) { 7324 return NO_ERROR; 7325 } 7326 mEnabled = false; 7327 7328 if (mEffect->suspended()) { 7329 return NO_ERROR; 7330 } 7331 7332 status_t status = mEffect->setEnabled(false); 7333 7334 sp<ThreadBase> thread = mEffect->thread().promote(); 7335 if (thread != 0) { 7336 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7337 } 7338 7339 return status; 7340} 7341 7342void AudioFlinger::EffectHandle::disconnect() 7343{ 7344 disconnect(true); 7345} 7346 7347void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7348{ 7349 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7350 if (mEffect == 0) { 7351 return; 7352 } 7353 mEffect->disconnect(this, unpinIfLast); 7354 7355 if (mHasControl && mEnabled) { 7356 sp<ThreadBase> thread = mEffect->thread().promote(); 7357 if (thread != 0) { 7358 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7359 } 7360 } 7361 7362 // release sp on module => module destructor can be called now 7363 mEffect.clear(); 7364 if (mClient != 0) { 7365 if (mCblk != NULL) { 7366 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7367 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7368 } 7369 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7370 // Client destructor must run with AudioFlinger mutex locked 7371 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7372 mClient.clear(); 7373 } 7374} 7375 7376status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7377 uint32_t cmdSize, 7378 void *pCmdData, 7379 uint32_t *replySize, 7380 void *pReplyData) 7381{ 7382// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7383// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7384 7385 // only get parameter command is permitted for applications not controlling the effect 7386 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7387 return INVALID_OPERATION; 7388 } 7389 if (mEffect == 0) return DEAD_OBJECT; 7390 if (mClient == 0) return INVALID_OPERATION; 7391 7392 // handle commands that are not forwarded transparently to effect engine 7393 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7394 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7395 // no risk to block the whole media server process or mixer threads is we are stuck here 7396 Mutex::Autolock _l(mCblk->lock); 7397 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7398 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7399 mCblk->serverIndex = 0; 7400 mCblk->clientIndex = 0; 7401 return BAD_VALUE; 7402 } 7403 status_t status = NO_ERROR; 7404 while (mCblk->serverIndex < mCblk->clientIndex) { 7405 int reply; 7406 uint32_t rsize = sizeof(int); 7407 int *p = (int *)(mBuffer + mCblk->serverIndex); 7408 int size = *p++; 7409 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7410 ALOGW("command(): invalid parameter block size"); 7411 break; 7412 } 7413 effect_param_t *param = (effect_param_t *)p; 7414 if (param->psize == 0 || param->vsize == 0) { 7415 ALOGW("command(): null parameter or value size"); 7416 mCblk->serverIndex += size; 7417 continue; 7418 } 7419 uint32_t psize = sizeof(effect_param_t) + 7420 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7421 param->vsize; 7422 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7423 psize, 7424 p, 7425 &rsize, 7426 &reply); 7427 // stop at first error encountered 7428 if (ret != NO_ERROR) { 7429 status = ret; 7430 *(int *)pReplyData = reply; 7431 break; 7432 } else if (reply != NO_ERROR) { 7433 *(int *)pReplyData = reply; 7434 break; 7435 } 7436 mCblk->serverIndex += size; 7437 } 7438 mCblk->serverIndex = 0; 7439 mCblk->clientIndex = 0; 7440 return status; 7441 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7442 *(int *)pReplyData = NO_ERROR; 7443 return enable(); 7444 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7445 *(int *)pReplyData = NO_ERROR; 7446 return disable(); 7447 } 7448 7449 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7450} 7451 7452void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7453{ 7454 ALOGV("setControl %p control %d", this, hasControl); 7455 7456 mHasControl = hasControl; 7457 mEnabled = enabled; 7458 7459 if (signal && mEffectClient != 0) { 7460 mEffectClient->controlStatusChanged(hasControl); 7461 } 7462} 7463 7464void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7465 uint32_t cmdSize, 7466 void *pCmdData, 7467 uint32_t replySize, 7468 void *pReplyData) 7469{ 7470 if (mEffectClient != 0) { 7471 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7472 } 7473} 7474 7475 7476 7477void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7478{ 7479 if (mEffectClient != 0) { 7480 mEffectClient->enableStatusChanged(enabled); 7481 } 7482} 7483 7484status_t AudioFlinger::EffectHandle::onTransact( 7485 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7486{ 7487 return BnEffect::onTransact(code, data, reply, flags); 7488} 7489 7490 7491void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7492{ 7493 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7494 7495 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7496 (mClient == 0) ? getpid_cached : mClient->pid(), 7497 mPriority, 7498 mHasControl, 7499 !locked, 7500 mCblk ? mCblk->clientIndex : 0, 7501 mCblk ? mCblk->serverIndex : 0 7502 ); 7503 7504 if (locked) { 7505 mCblk->lock.unlock(); 7506 } 7507} 7508 7509#undef LOG_TAG 7510#define LOG_TAG "AudioFlinger::EffectChain" 7511 7512AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7513 int sessionId) 7514 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7515 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7516 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7517{ 7518 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7519 if (thread == NULL) { 7520 return; 7521 } 7522 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7523 thread->frameCount(); 7524} 7525 7526AudioFlinger::EffectChain::~EffectChain() 7527{ 7528 if (mOwnInBuffer) { 7529 delete mInBuffer; 7530 } 7531 7532} 7533 7534// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7535sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7536{ 7537 size_t size = mEffects.size(); 7538 7539 for (size_t i = 0; i < size; i++) { 7540 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7541 return mEffects[i]; 7542 } 7543 } 7544 return 0; 7545} 7546 7547// getEffectFromId_l() must be called with ThreadBase::mLock held 7548sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7549{ 7550 size_t size = mEffects.size(); 7551 7552 for (size_t i = 0; i < size; i++) { 7553 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7554 if (id == 0 || mEffects[i]->id() == id) { 7555 return mEffects[i]; 7556 } 7557 } 7558 return 0; 7559} 7560 7561// getEffectFromType_l() must be called with ThreadBase::mLock held 7562sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7563 const effect_uuid_t *type) 7564{ 7565 size_t size = mEffects.size(); 7566 7567 for (size_t i = 0; i < size; i++) { 7568 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7569 return mEffects[i]; 7570 } 7571 } 7572 return 0; 7573} 7574 7575// Must be called with EffectChain::mLock locked 7576void AudioFlinger::EffectChain::process_l() 7577{ 7578 sp<ThreadBase> thread = mThread.promote(); 7579 if (thread == 0) { 7580 ALOGW("process_l(): cannot promote mixer thread"); 7581 return; 7582 } 7583 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7584 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7585 // always process effects unless no more tracks are on the session and the effect tail 7586 // has been rendered 7587 bool doProcess = true; 7588 if (!isGlobalSession) { 7589 bool tracksOnSession = (trackCnt() != 0); 7590 7591 if (!tracksOnSession && mTailBufferCount == 0) { 7592 doProcess = false; 7593 } 7594 7595 if (activeTrackCnt() == 0) { 7596 // if no track is active and the effect tail has not been rendered, 7597 // the input buffer must be cleared here as the mixer process will not do it 7598 if (tracksOnSession || mTailBufferCount > 0) { 7599 size_t numSamples = thread->frameCount() * thread->channelCount(); 7600 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7601 if (mTailBufferCount > 0) { 7602 mTailBufferCount--; 7603 } 7604 } 7605 } 7606 } 7607 7608 size_t size = mEffects.size(); 7609 if (doProcess) { 7610 for (size_t i = 0; i < size; i++) { 7611 mEffects[i]->process(); 7612 } 7613 } 7614 for (size_t i = 0; i < size; i++) { 7615 mEffects[i]->updateState(); 7616 } 7617} 7618 7619// addEffect_l() must be called with PlaybackThread::mLock held 7620status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7621{ 7622 effect_descriptor_t desc = effect->desc(); 7623 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7624 7625 Mutex::Autolock _l(mLock); 7626 effect->setChain(this); 7627 sp<ThreadBase> thread = mThread.promote(); 7628 if (thread == 0) { 7629 return NO_INIT; 7630 } 7631 effect->setThread(thread); 7632 7633 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7634 // Auxiliary effects are inserted at the beginning of mEffects vector as 7635 // they are processed first and accumulated in chain input buffer 7636 mEffects.insertAt(effect, 0); 7637 7638 // the input buffer for auxiliary effect contains mono samples in 7639 // 32 bit format. This is to avoid saturation in AudoMixer 7640 // accumulation stage. Saturation is done in EffectModule::process() before 7641 // calling the process in effect engine 7642 size_t numSamples = thread->frameCount(); 7643 int32_t *buffer = new int32_t[numSamples]; 7644 memset(buffer, 0, numSamples * sizeof(int32_t)); 7645 effect->setInBuffer((int16_t *)buffer); 7646 // auxiliary effects output samples to chain input buffer for further processing 7647 // by insert effects 7648 effect->setOutBuffer(mInBuffer); 7649 } else { 7650 // Insert effects are inserted at the end of mEffects vector as they are processed 7651 // after track and auxiliary effects. 7652 // Insert effect order as a function of indicated preference: 7653 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7654 // another effect is present 7655 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7656 // last effect claiming first position 7657 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7658 // first effect claiming last position 7659 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7660 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7661 // already present 7662 7663 size_t size = mEffects.size(); 7664 size_t idx_insert = size; 7665 ssize_t idx_insert_first = -1; 7666 ssize_t idx_insert_last = -1; 7667 7668 for (size_t i = 0; i < size; i++) { 7669 effect_descriptor_t d = mEffects[i]->desc(); 7670 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7671 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7672 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7673 // check invalid effect chaining combinations 7674 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7675 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7676 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7677 return INVALID_OPERATION; 7678 } 7679 // remember position of first insert effect and by default 7680 // select this as insert position for new effect 7681 if (idx_insert == size) { 7682 idx_insert = i; 7683 } 7684 // remember position of last insert effect claiming 7685 // first position 7686 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7687 idx_insert_first = i; 7688 } 7689 // remember position of first insert effect claiming 7690 // last position 7691 if (iPref == EFFECT_FLAG_INSERT_LAST && 7692 idx_insert_last == -1) { 7693 idx_insert_last = i; 7694 } 7695 } 7696 } 7697 7698 // modify idx_insert from first position if needed 7699 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7700 if (idx_insert_last != -1) { 7701 idx_insert = idx_insert_last; 7702 } else { 7703 idx_insert = size; 7704 } 7705 } else { 7706 if (idx_insert_first != -1) { 7707 idx_insert = idx_insert_first + 1; 7708 } 7709 } 7710 7711 // always read samples from chain input buffer 7712 effect->setInBuffer(mInBuffer); 7713 7714 // if last effect in the chain, output samples to chain 7715 // output buffer, otherwise to chain input buffer 7716 if (idx_insert == size) { 7717 if (idx_insert != 0) { 7718 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7719 mEffects[idx_insert-1]->configure(); 7720 } 7721 effect->setOutBuffer(mOutBuffer); 7722 } else { 7723 effect->setOutBuffer(mInBuffer); 7724 } 7725 mEffects.insertAt(effect, idx_insert); 7726 7727 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7728 } 7729 effect->configure(); 7730 return NO_ERROR; 7731} 7732 7733// removeEffect_l() must be called with PlaybackThread::mLock held 7734size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7735{ 7736 Mutex::Autolock _l(mLock); 7737 size_t size = mEffects.size(); 7738 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7739 7740 for (size_t i = 0; i < size; i++) { 7741 if (effect == mEffects[i]) { 7742 // calling stop here will remove pre-processing effect from the audio HAL. 7743 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7744 // the middle of a read from audio HAL 7745 if (mEffects[i]->state() == EffectModule::ACTIVE || 7746 mEffects[i]->state() == EffectModule::STOPPING) { 7747 mEffects[i]->stop(); 7748 } 7749 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7750 delete[] effect->inBuffer(); 7751 } else { 7752 if (i == size - 1 && i != 0) { 7753 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7754 mEffects[i - 1]->configure(); 7755 } 7756 } 7757 mEffects.removeAt(i); 7758 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7759 break; 7760 } 7761 } 7762 7763 return mEffects.size(); 7764} 7765 7766// setDevice_l() must be called with PlaybackThread::mLock held 7767void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7768{ 7769 size_t size = mEffects.size(); 7770 for (size_t i = 0; i < size; i++) { 7771 mEffects[i]->setDevice(device); 7772 } 7773} 7774 7775// setMode_l() must be called with PlaybackThread::mLock held 7776void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7777{ 7778 size_t size = mEffects.size(); 7779 for (size_t i = 0; i < size; i++) { 7780 mEffects[i]->setMode(mode); 7781 } 7782} 7783 7784// setVolume_l() must be called with PlaybackThread::mLock held 7785bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7786{ 7787 uint32_t newLeft = *left; 7788 uint32_t newRight = *right; 7789 bool hasControl = false; 7790 int ctrlIdx = -1; 7791 size_t size = mEffects.size(); 7792 7793 // first update volume controller 7794 for (size_t i = size; i > 0; i--) { 7795 if (mEffects[i - 1]->isProcessEnabled() && 7796 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7797 ctrlIdx = i - 1; 7798 hasControl = true; 7799 break; 7800 } 7801 } 7802 7803 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7804 if (hasControl) { 7805 *left = mNewLeftVolume; 7806 *right = mNewRightVolume; 7807 } 7808 return hasControl; 7809 } 7810 7811 mVolumeCtrlIdx = ctrlIdx; 7812 mLeftVolume = newLeft; 7813 mRightVolume = newRight; 7814 7815 // second get volume update from volume controller 7816 if (ctrlIdx >= 0) { 7817 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7818 mNewLeftVolume = newLeft; 7819 mNewRightVolume = newRight; 7820 } 7821 // then indicate volume to all other effects in chain. 7822 // Pass altered volume to effects before volume controller 7823 // and requested volume to effects after controller 7824 uint32_t lVol = newLeft; 7825 uint32_t rVol = newRight; 7826 7827 for (size_t i = 0; i < size; i++) { 7828 if ((int)i == ctrlIdx) continue; 7829 // this also works for ctrlIdx == -1 when there is no volume controller 7830 if ((int)i > ctrlIdx) { 7831 lVol = *left; 7832 rVol = *right; 7833 } 7834 mEffects[i]->setVolume(&lVol, &rVol, false); 7835 } 7836 *left = newLeft; 7837 *right = newRight; 7838 7839 return hasControl; 7840} 7841 7842status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7843{ 7844 const size_t SIZE = 256; 7845 char buffer[SIZE]; 7846 String8 result; 7847 7848 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7849 result.append(buffer); 7850 7851 bool locked = tryLock(mLock); 7852 // failed to lock - AudioFlinger is probably deadlocked 7853 if (!locked) { 7854 result.append("\tCould not lock mutex:\n"); 7855 } 7856 7857 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7858 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7859 mEffects.size(), 7860 (uint32_t)mInBuffer, 7861 (uint32_t)mOutBuffer, 7862 mActiveTrackCnt); 7863 result.append(buffer); 7864 write(fd, result.string(), result.size()); 7865 7866 for (size_t i = 0; i < mEffects.size(); ++i) { 7867 sp<EffectModule> effect = mEffects[i]; 7868 if (effect != 0) { 7869 effect->dump(fd, args); 7870 } 7871 } 7872 7873 if (locked) { 7874 mLock.unlock(); 7875 } 7876 7877 return NO_ERROR; 7878} 7879 7880// must be called with ThreadBase::mLock held 7881void AudioFlinger::EffectChain::setEffectSuspended_l( 7882 const effect_uuid_t *type, bool suspend) 7883{ 7884 sp<SuspendedEffectDesc> desc; 7885 // use effect type UUID timelow as key as there is no real risk of identical 7886 // timeLow fields among effect type UUIDs. 7887 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7888 if (suspend) { 7889 if (index >= 0) { 7890 desc = mSuspendedEffects.valueAt(index); 7891 } else { 7892 desc = new SuspendedEffectDesc(); 7893 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7894 mSuspendedEffects.add(type->timeLow, desc); 7895 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7896 } 7897 if (desc->mRefCount++ == 0) { 7898 sp<EffectModule> effect = getEffectIfEnabled(type); 7899 if (effect != 0) { 7900 desc->mEffect = effect; 7901 effect->setSuspended(true); 7902 effect->setEnabled(false); 7903 } 7904 } 7905 } else { 7906 if (index < 0) { 7907 return; 7908 } 7909 desc = mSuspendedEffects.valueAt(index); 7910 if (desc->mRefCount <= 0) { 7911 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7912 desc->mRefCount = 1; 7913 } 7914 if (--desc->mRefCount == 0) { 7915 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7916 if (desc->mEffect != 0) { 7917 sp<EffectModule> effect = desc->mEffect.promote(); 7918 if (effect != 0) { 7919 effect->setSuspended(false); 7920 sp<EffectHandle> handle = effect->controlHandle(); 7921 if (handle != 0) { 7922 effect->setEnabled(handle->enabled()); 7923 } 7924 } 7925 desc->mEffect.clear(); 7926 } 7927 mSuspendedEffects.removeItemsAt(index); 7928 } 7929 } 7930} 7931 7932// must be called with ThreadBase::mLock held 7933void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7934{ 7935 sp<SuspendedEffectDesc> desc; 7936 7937 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7938 if (suspend) { 7939 if (index >= 0) { 7940 desc = mSuspendedEffects.valueAt(index); 7941 } else { 7942 desc = new SuspendedEffectDesc(); 7943 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7944 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7945 } 7946 if (desc->mRefCount++ == 0) { 7947 Vector< sp<EffectModule> > effects; 7948 getSuspendEligibleEffects(effects); 7949 for (size_t i = 0; i < effects.size(); i++) { 7950 setEffectSuspended_l(&effects[i]->desc().type, true); 7951 } 7952 } 7953 } else { 7954 if (index < 0) { 7955 return; 7956 } 7957 desc = mSuspendedEffects.valueAt(index); 7958 if (desc->mRefCount <= 0) { 7959 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7960 desc->mRefCount = 1; 7961 } 7962 if (--desc->mRefCount == 0) { 7963 Vector<const effect_uuid_t *> types; 7964 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7965 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7966 continue; 7967 } 7968 types.add(&mSuspendedEffects.valueAt(i)->mType); 7969 } 7970 for (size_t i = 0; i < types.size(); i++) { 7971 setEffectSuspended_l(types[i], false); 7972 } 7973 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7974 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7975 } 7976 } 7977} 7978 7979 7980// The volume effect is used for automated tests only 7981#ifndef OPENSL_ES_H_ 7982static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7983 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7984const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7985#endif //OPENSL_ES_H_ 7986 7987bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7988{ 7989 // auxiliary effects and visualizer are never suspended on output mix 7990 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7991 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7992 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7993 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7994 return false; 7995 } 7996 return true; 7997} 7998 7999void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8000{ 8001 effects.clear(); 8002 for (size_t i = 0; i < mEffects.size(); i++) { 8003 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8004 effects.add(mEffects[i]); 8005 } 8006 } 8007} 8008 8009sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8010 const effect_uuid_t *type) 8011{ 8012 sp<EffectModule> effect = getEffectFromType_l(type); 8013 return effect != 0 && effect->isEnabled() ? effect : 0; 8014} 8015 8016void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8017 bool enabled) 8018{ 8019 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8020 if (enabled) { 8021 if (index < 0) { 8022 // if the effect is not suspend check if all effects are suspended 8023 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8024 if (index < 0) { 8025 return; 8026 } 8027 if (!isEffectEligibleForSuspend(effect->desc())) { 8028 return; 8029 } 8030 setEffectSuspended_l(&effect->desc().type, enabled); 8031 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8032 if (index < 0) { 8033 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8034 return; 8035 } 8036 } 8037 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8038 effect->desc().type.timeLow); 8039 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8040 // if effect is requested to suspended but was not yet enabled, supend it now. 8041 if (desc->mEffect == 0) { 8042 desc->mEffect = effect; 8043 effect->setEnabled(false); 8044 effect->setSuspended(true); 8045 } 8046 } else { 8047 if (index < 0) { 8048 return; 8049 } 8050 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8051 effect->desc().type.timeLow); 8052 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8053 desc->mEffect.clear(); 8054 effect->setSuspended(false); 8055 } 8056} 8057 8058#undef LOG_TAG 8059#define LOG_TAG "AudioFlinger" 8060 8061// ---------------------------------------------------------------------------- 8062 8063status_t AudioFlinger::onTransact( 8064 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8065{ 8066 return BnAudioFlinger::onTransact(code, data, reply, flags); 8067} 8068 8069}; // namespace android 8070