AudioFlinger.cpp revision b81cc8c6f3eec9edb255ea99b6a6f243585b1e38
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 // MIXER || DUPLICATING 1997 Vector< sp<Track> > tracksToRemove; 1998 1999 // DIRECT 2000 sp<Track> trackToRemove; 2001 2002 standbyTime = systemTime(); 2003 mixBufferSize = mFrameCount * mFrameSize; 2004 2005 // MIXER 2006 // FIXME: Relaxed timing because of a certain device that can't meet latency 2007 // Should be reduced to 2x after the vendor fixes the driver issue 2008 // increase threshold again due to low power audio mode. The way this warning threshold is 2009 // calculated and its usefulness should be reconsidered anyway. 2010 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2011 nsecs_t lastWarning = 0; 2012if (mType == MIXER) { 2013 longStandbyExit = false; 2014} 2015 2016 // DUPLICATING 2017 // FIXME could this be made local to while loop? 2018 writeFrames = 0; 2019 2020 activeSleepTime = activeSleepTimeUs(); 2021 idleSleepTime = idleSleepTimeUs(); 2022 sleepTime = idleSleepTime; 2023 2024if (mType == MIXER) { 2025 sleepTimeShift = 0; 2026} 2027 2028 // MIXER 2029 CpuStats cpuStats; 2030 2031 // DIRECT 2032if (mType == DIRECT) { 2033 // use shorter standby delay as on normal output to release 2034 // hardware resources as soon as possible 2035 standbyDelay = microseconds(activeSleepTime*2); 2036} 2037 2038 acquireWakeLock(); 2039 2040 while (!exitPending()) 2041 { 2042if (mType == MIXER) { 2043 cpuStats.sample(); 2044} 2045 2046 Vector< sp<EffectChain> > effectChains; 2047 2048 processConfigEvents(); 2049 2050if (mType == DIRECT) { 2051 activeTrack.clear(); 2052} 2053 2054 mixerStatus = MIXER_IDLE; 2055 { // scope for mLock 2056 2057 Mutex::Autolock _l(mLock); 2058 2059 if (checkForNewParameters_l()) { 2060 mixBufferSize = mFrameCount * mFrameSize; 2061 2062if (mType == MIXER) { 2063 // FIXME: Relaxed timing because of a certain device that can't meet latency 2064 // Should be reduced to 2x after the vendor fixes the driver issue 2065 // increase threshold again due to low power audio mode. The way this warning 2066 // threshold is calculated and its usefulness should be reconsidered anyway. 2067 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2068} 2069 2070 updateWaitTime_l(); 2071 2072 activeSleepTime = activeSleepTimeUs(); 2073 idleSleepTime = idleSleepTimeUs(); 2074 2075if (mType == DIRECT) { 2076 standbyDelay = microseconds(activeSleepTime*2); 2077} 2078 2079 } 2080 2081if (mType == DUPLICATING) { 2082#if 0 // see earlier FIXME 2083 // Now that this is a field instead of local variable, 2084 // clear it so it is empty the first time through the loop, 2085 // and later an assignment could combine the clear with the loop below 2086 outputTracks.clear(); 2087#endif 2088 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2089 outputTracks.add(mOutputTracks[i]); 2090 } 2091} 2092 2093 // put audio hardware into standby after short delay 2094 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2095 mSuspended > 0)) { 2096 if (!mStandby) { 2097 2098 threadLoop_standby(); 2099 2100 mStandby = true; 2101 mBytesWritten = 0; 2102 } 2103 2104 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2105 // we're about to wait, flush the binder command buffer 2106 IPCThreadState::self()->flushCommands(); 2107 2108if (mType == DUPLICATING) { 2109 outputTracks.clear(); 2110} 2111 2112 if (exitPending()) break; 2113 2114 releaseWakeLock_l(); 2115 // wait until we have something to do... 2116 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2117 mWaitWorkCV.wait(mLock); 2118 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2119 acquireWakeLock_l(); 2120 2121if (mType == MIXER || mType == DUPLICATING) { 2122 mPrevMixerStatus = MIXER_IDLE; 2123} 2124 2125 checkSilentMode_l(); 2126 2127if (mType == MIXER || mType == DUPLICATING) { 2128 standbyTime = systemTime() + mStandbyTimeInNsecs; 2129} 2130 2131if (mType == DIRECT) { 2132 standbyTime = systemTime() + standbyDelay; 2133} 2134 2135 sleepTime = idleSleepTime; 2136 2137if (mType == MIXER) { 2138 sleepTimeShift = 0; 2139} 2140 2141 continue; 2142 } 2143 } 2144 2145// FIXME merge these 2146if (mType == MIXER || mType == DUPLICATING) { 2147 mixerStatus = prepareTracks_l(&tracksToRemove); 2148} 2149if (mType == DIRECT) { 2150 mixerStatus = threadLoop_prepareTracks_l(trackToRemove); 2151 // see FIXME in AudioFlinger.h 2152 if (mixerStatus == MIXER_CONTINUE) { 2153 continue; 2154 } 2155} 2156 2157 // prevent any changes in effect chain list and in each effect chain 2158 // during mixing and effect process as the audio buffers could be deleted 2159 // or modified if an effect is created or deleted 2160 lockEffectChains_l(effectChains); 2161 } 2162 2163if (mType == DIRECT) { 2164 // For DirectOutputThread, this test is equivalent to "activeTrack != 0" 2165} 2166 2167 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2168 threadLoop_mix(); 2169 } else { 2170 threadLoop_sleepTime(); 2171 } 2172 2173 if (mSuspended > 0) { 2174 sleepTime = suspendSleepTimeUs(); 2175 } 2176 2177 // only process effects if we're going to write 2178 if (sleepTime == 0) { 2179 2180 if (mixerStatus == MIXER_TRACKS_READY) { 2181 2182 // Non-trivial for DIRECT only 2183 applyVolume(); 2184 2185 } 2186 2187 for (size_t i = 0; i < effectChains.size(); i ++) { 2188 effectChains[i]->process_l(); 2189 } 2190 } 2191 2192 // enable changes in effect chain 2193 unlockEffectChains(effectChains); 2194 2195 // sleepTime == 0 means we must write to audio hardware 2196 if (sleepTime == 0) { 2197 2198 threadLoop_write(); 2199 2200if (mType == MIXER) { 2201 // write blocked detection 2202 nsecs_t now = systemTime(); 2203 nsecs_t delta = now - mLastWriteTime; 2204 if (!mStandby && delta > maxPeriod) { 2205 mNumDelayedWrites++; 2206 if ((now - lastWarning) > kWarningThrottleNs) { 2207 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2208 ns2ms(delta), mNumDelayedWrites, this); 2209 lastWarning = now; 2210 } 2211 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2212 // a different threshold. Or completely removed for what it is worth anyway... 2213 if (mStandby) { 2214 longStandbyExit = true; 2215 } 2216 } 2217} 2218 2219 mStandby = false; 2220 } else { 2221 usleep(sleepTime); 2222 } 2223 2224 // finally let go of removed track(s), without the lock held 2225 // since we can't guarantee the destructors won't acquire that 2226 // same lock. 2227 2228// FIXME merge these 2229if (mType == MIXER) { 2230 tracksToRemove.clear(); 2231} 2232if (mType == DIRECT) { 2233 trackToRemove.clear(); 2234 activeTrack.clear(); 2235} 2236if (mType == DUPLICATING) { 2237 tracksToRemove.clear(); 2238 outputTracks.clear(); 2239} 2240 2241 // Effect chains will be actually deleted here if they were removed from 2242 // mEffectChains list during mixing or effects processing 2243 effectChains.clear(); 2244 2245 // FIXME Note that the above .clear() is no longer necessary since effectChains 2246 // is now local to this block, but will keep it for now (at least until merge done). 2247 } 2248 2249if (mType == MIXER || mType == DIRECT) { 2250 // put output stream into standby mode 2251 if (!mStandby) { 2252 mOutput->stream->common.standby(&mOutput->stream->common); 2253 } 2254} 2255if (mType == DUPLICATING) { 2256 // for DuplicatingThread, standby mode is handled by the outputTracks 2257} 2258 2259 releaseWakeLock(); 2260 2261 ALOGV("Thread %p type %d exiting", this, mType); 2262 return false; 2263} 2264 2265// shared by MIXER and DIRECT, overridden by DUPLICATING 2266void AudioFlinger::PlaybackThread::threadLoop_write() 2267{ 2268 // FIXME rewrite to reduce number of system calls 2269 mLastWriteTime = systemTime(); 2270 mInWrite = true; 2271 mBytesWritten += mixBufferSize; 2272 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2273 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2274 mNumWrites++; 2275 mInWrite = false; 2276} 2277 2278// shared by MIXER and DIRECT, overridden by DUPLICATING 2279void AudioFlinger::PlaybackThread::threadLoop_standby() 2280{ 2281 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2282 mOutput->stream->common.standby(&mOutput->stream->common); 2283} 2284 2285void AudioFlinger::MixerThread::threadLoop_mix() 2286{ 2287 // obtain the presentation timestamp of the next output buffer 2288 int64_t pts; 2289 status_t status = INVALID_OPERATION; 2290 2291 if (NULL != mOutput->stream->get_next_write_timestamp) { 2292 status = mOutput->stream->get_next_write_timestamp( 2293 mOutput->stream, &pts); 2294 } 2295 2296 if (status != NO_ERROR) { 2297 pts = AudioBufferProvider::kInvalidPTS; 2298 } 2299 2300 // mix buffers... 2301 mAudioMixer->process(pts); 2302 // increase sleep time progressively when application underrun condition clears. 2303 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2304 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2305 // such that we would underrun the audio HAL. 2306 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2307 sleepTimeShift--; 2308 } 2309 sleepTime = 0; 2310 standbyTime = systemTime() + mStandbyTimeInNsecs; 2311 //TODO: delay standby when effects have a tail 2312} 2313 2314void AudioFlinger::MixerThread::threadLoop_sleepTime() 2315{ 2316 // If no tracks are ready, sleep once for the duration of an output 2317 // buffer size, then write 0s to the output 2318 if (sleepTime == 0) { 2319 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2320 sleepTime = activeSleepTime >> sleepTimeShift; 2321 if (sleepTime < kMinThreadSleepTimeUs) { 2322 sleepTime = kMinThreadSleepTimeUs; 2323 } 2324 // reduce sleep time in case of consecutive application underruns to avoid 2325 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2326 // duration we would end up writing less data than needed by the audio HAL if 2327 // the condition persists. 2328 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2329 sleepTimeShift++; 2330 } 2331 } else { 2332 sleepTime = idleSleepTime; 2333 } 2334 } else if (mBytesWritten != 0 || 2335 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2336 memset (mMixBuffer, 0, mixBufferSize); 2337 sleepTime = 0; 2338 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2339 } 2340 // TODO add standby time extension fct of effect tail 2341} 2342 2343// prepareTracks_l() must be called with ThreadBase::mLock held 2344AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2345 Vector< sp<Track> > *tracksToRemove) 2346{ 2347 2348 mixer_state mixerStatus = MIXER_IDLE; 2349 // find out which tracks need to be processed 2350 size_t count = mActiveTracks.size(); 2351 size_t mixedTracks = 0; 2352 size_t tracksWithEffect = 0; 2353 2354 float masterVolume = mMasterVolume; 2355 bool masterMute = mMasterMute; 2356 2357 if (masterMute) { 2358 masterVolume = 0; 2359 } 2360 // Delegate master volume control to effect in output mix effect chain if needed 2361 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2362 if (chain != 0) { 2363 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2364 chain->setVolume_l(&v, &v); 2365 masterVolume = (float)((v + (1 << 23)) >> 24); 2366 chain.clear(); 2367 } 2368 2369 for (size_t i=0 ; i<count ; i++) { 2370 sp<Track> t = mActiveTracks[i].promote(); 2371 if (t == 0) continue; 2372 2373 // this const just means the local variable doesn't change 2374 Track* const track = t.get(); 2375 audio_track_cblk_t* cblk = track->cblk(); 2376 2377 // The first time a track is added we wait 2378 // for all its buffers to be filled before processing it 2379 int name = track->name(); 2380 // make sure that we have enough frames to mix one full buffer. 2381 // enforce this condition only once to enable draining the buffer in case the client 2382 // app does not call stop() and relies on underrun to stop: 2383 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2384 // during last round 2385 uint32_t minFrames = 1; 2386 if (!track->isStopped() && !track->isPausing() && 2387 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2388 if (t->sampleRate() == (int)mSampleRate) { 2389 minFrames = mFrameCount; 2390 } else { 2391 // +1 for rounding and +1 for additional sample needed for interpolation 2392 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2393 // add frames already consumed but not yet released by the resampler 2394 // because cblk->framesReady() will include these frames 2395 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2396 // the minimum track buffer size is normally twice the number of frames necessary 2397 // to fill one buffer and the resampler should not leave more than one buffer worth 2398 // of unreleased frames after each pass, but just in case... 2399 ALOG_ASSERT(minFrames <= cblk->frameCount); 2400 } 2401 } 2402 if ((track->framesReady() >= minFrames) && track->isReady() && 2403 !track->isPaused() && !track->isTerminated()) 2404 { 2405 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2406 2407 mixedTracks++; 2408 2409 // track->mainBuffer() != mMixBuffer means there is an effect chain 2410 // connected to the track 2411 chain.clear(); 2412 if (track->mainBuffer() != mMixBuffer) { 2413 chain = getEffectChain_l(track->sessionId()); 2414 // Delegate volume control to effect in track effect chain if needed 2415 if (chain != 0) { 2416 tracksWithEffect++; 2417 } else { 2418 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2419 name, track->sessionId()); 2420 } 2421 } 2422 2423 2424 int param = AudioMixer::VOLUME; 2425 if (track->mFillingUpStatus == Track::FS_FILLED) { 2426 // no ramp for the first volume setting 2427 track->mFillingUpStatus = Track::FS_ACTIVE; 2428 if (track->mState == TrackBase::RESUMING) { 2429 track->mState = TrackBase::ACTIVE; 2430 param = AudioMixer::RAMP_VOLUME; 2431 } 2432 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2433 } else if (cblk->server != 0) { 2434 // If the track is stopped before the first frame was mixed, 2435 // do not apply ramp 2436 param = AudioMixer::RAMP_VOLUME; 2437 } 2438 2439 // compute volume for this track 2440 uint32_t vl, vr, va; 2441 if (track->isMuted() || track->isPausing() || 2442 mStreamTypes[track->streamType()].mute) { 2443 vl = vr = va = 0; 2444 if (track->isPausing()) { 2445 track->setPaused(); 2446 } 2447 } else { 2448 2449 // read original volumes with volume control 2450 float typeVolume = mStreamTypes[track->streamType()].volume; 2451 float v = masterVolume * typeVolume; 2452 uint32_t vlr = cblk->getVolumeLR(); 2453 vl = vlr & 0xFFFF; 2454 vr = vlr >> 16; 2455 // track volumes come from shared memory, so can't be trusted and must be clamped 2456 if (vl > MAX_GAIN_INT) { 2457 ALOGV("Track left volume out of range: %04X", vl); 2458 vl = MAX_GAIN_INT; 2459 } 2460 if (vr > MAX_GAIN_INT) { 2461 ALOGV("Track right volume out of range: %04X", vr); 2462 vr = MAX_GAIN_INT; 2463 } 2464 // now apply the master volume and stream type volume 2465 vl = (uint32_t)(v * vl) << 12; 2466 vr = (uint32_t)(v * vr) << 12; 2467 // assuming master volume and stream type volume each go up to 1.0, 2468 // vl and vr are now in 8.24 format 2469 2470 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2471 // send level comes from shared memory and so may be corrupt 2472 if (sendLevel > MAX_GAIN_INT) { 2473 ALOGV("Track send level out of range: %04X", sendLevel); 2474 sendLevel = MAX_GAIN_INT; 2475 } 2476 va = (uint32_t)(v * sendLevel); 2477 } 2478 // Delegate volume control to effect in track effect chain if needed 2479 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2480 // Do not ramp volume if volume is controlled by effect 2481 param = AudioMixer::VOLUME; 2482 track->mHasVolumeController = true; 2483 } else { 2484 // force no volume ramp when volume controller was just disabled or removed 2485 // from effect chain to avoid volume spike 2486 if (track->mHasVolumeController) { 2487 param = AudioMixer::VOLUME; 2488 } 2489 track->mHasVolumeController = false; 2490 } 2491 2492 // Convert volumes from 8.24 to 4.12 format 2493 // This additional clamping is needed in case chain->setVolume_l() overshot 2494 vl = (vl + (1 << 11)) >> 12; 2495 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2496 vr = (vr + (1 << 11)) >> 12; 2497 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2498 2499 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2500 2501 // XXX: these things DON'T need to be done each time 2502 mAudioMixer->setBufferProvider(name, track); 2503 mAudioMixer->enable(name); 2504 2505 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2506 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2507 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2508 mAudioMixer->setParameter( 2509 name, 2510 AudioMixer::TRACK, 2511 AudioMixer::FORMAT, (void *)track->format()); 2512 mAudioMixer->setParameter( 2513 name, 2514 AudioMixer::TRACK, 2515 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2516 mAudioMixer->setParameter( 2517 name, 2518 AudioMixer::RESAMPLE, 2519 AudioMixer::SAMPLE_RATE, 2520 (void *)(cblk->sampleRate)); 2521 mAudioMixer->setParameter( 2522 name, 2523 AudioMixer::TRACK, 2524 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2525 mAudioMixer->setParameter( 2526 name, 2527 AudioMixer::TRACK, 2528 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2529 2530 // reset retry count 2531 track->mRetryCount = kMaxTrackRetries; 2532 // If one track is ready, set the mixer ready if: 2533 // - the mixer was not ready during previous round OR 2534 // - no other track is not ready 2535 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2536 mixerStatus != MIXER_TRACKS_ENABLED) { 2537 mixerStatus = MIXER_TRACKS_READY; 2538 } 2539 } else { 2540 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2541 if (track->isStopped()) { 2542 track->reset(); 2543 } 2544 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2545 // We have consumed all the buffers of this track. 2546 // Remove it from the list of active tracks. 2547 tracksToRemove->add(track); 2548 } else { 2549 // No buffers for this track. Give it a few chances to 2550 // fill a buffer, then remove it from active list. 2551 if (--(track->mRetryCount) <= 0) { 2552 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2553 tracksToRemove->add(track); 2554 // indicate to client process that the track was disabled because of underrun 2555 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2556 // If one track is not ready, mark the mixer also not ready if: 2557 // - the mixer was ready during previous round OR 2558 // - no other track is ready 2559 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2560 mixerStatus != MIXER_TRACKS_READY) { 2561 mixerStatus = MIXER_TRACKS_ENABLED; 2562 } 2563 } 2564 mAudioMixer->disable(name); 2565 } 2566 } 2567 2568 // remove all the tracks that need to be... 2569 count = tracksToRemove->size(); 2570 if (CC_UNLIKELY(count)) { 2571 for (size_t i=0 ; i<count ; i++) { 2572 const sp<Track>& track = tracksToRemove->itemAt(i); 2573 mActiveTracks.remove(track); 2574 if (track->mainBuffer() != mMixBuffer) { 2575 chain = getEffectChain_l(track->sessionId()); 2576 if (chain != 0) { 2577 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2578 chain->decActiveTrackCnt(); 2579 } 2580 } 2581 if (track->isTerminated()) { 2582 removeTrack_l(track); 2583 } 2584 } 2585 } 2586 2587 // mix buffer must be cleared if all tracks are connected to an 2588 // effect chain as in this case the mixer will not write to 2589 // mix buffer and track effects will accumulate into it 2590 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2591 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2592 } 2593 2594 mPrevMixerStatus = mixerStatus; 2595 return mixerStatus; 2596} 2597 2598void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2599{ 2600 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2601 this, streamType, mTracks.size()); 2602 Mutex::Autolock _l(mLock); 2603 2604 size_t size = mTracks.size(); 2605 for (size_t i = 0; i < size; i++) { 2606 sp<Track> t = mTracks[i]; 2607 if (t->streamType() == streamType) { 2608 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2609 t->mCblk->cv.signal(); 2610 } 2611 } 2612} 2613 2614void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2615{ 2616 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2617 this, streamType, valid); 2618 Mutex::Autolock _l(mLock); 2619 2620 mStreamTypes[streamType].valid = valid; 2621} 2622 2623// getTrackName_l() must be called with ThreadBase::mLock held 2624int AudioFlinger::MixerThread::getTrackName_l() 2625{ 2626 return mAudioMixer->getTrackName(); 2627} 2628 2629// deleteTrackName_l() must be called with ThreadBase::mLock held 2630void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2631{ 2632 ALOGV("remove track (%d) and delete from mixer", name); 2633 mAudioMixer->deleteTrackName(name); 2634} 2635 2636// checkForNewParameters_l() must be called with ThreadBase::mLock held 2637bool AudioFlinger::MixerThread::checkForNewParameters_l() 2638{ 2639 bool reconfig = false; 2640 2641 while (!mNewParameters.isEmpty()) { 2642 status_t status = NO_ERROR; 2643 String8 keyValuePair = mNewParameters[0]; 2644 AudioParameter param = AudioParameter(keyValuePair); 2645 int value; 2646 2647 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2648 reconfig = true; 2649 } 2650 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2651 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2652 status = BAD_VALUE; 2653 } else { 2654 reconfig = true; 2655 } 2656 } 2657 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2658 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2659 status = BAD_VALUE; 2660 } else { 2661 reconfig = true; 2662 } 2663 } 2664 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2665 // do not accept frame count changes if tracks are open as the track buffer 2666 // size depends on frame count and correct behavior would not be guaranteed 2667 // if frame count is changed after track creation 2668 if (!mTracks.isEmpty()) { 2669 status = INVALID_OPERATION; 2670 } else { 2671 reconfig = true; 2672 } 2673 } 2674 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2675 // when changing the audio output device, call addBatteryData to notify 2676 // the change 2677 if ((int)mDevice != value) { 2678 uint32_t params = 0; 2679 // check whether speaker is on 2680 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2681 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2682 } 2683 2684 int deviceWithoutSpeaker 2685 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2686 // check if any other device (except speaker) is on 2687 if (value & deviceWithoutSpeaker ) { 2688 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2689 } 2690 2691 if (params != 0) { 2692 addBatteryData(params); 2693 } 2694 } 2695 2696 // forward device change to effects that have requested to be 2697 // aware of attached audio device. 2698 mDevice = (uint32_t)value; 2699 for (size_t i = 0; i < mEffectChains.size(); i++) { 2700 mEffectChains[i]->setDevice_l(mDevice); 2701 } 2702 } 2703 2704 if (status == NO_ERROR) { 2705 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2706 keyValuePair.string()); 2707 if (!mStandby && status == INVALID_OPERATION) { 2708 mOutput->stream->common.standby(&mOutput->stream->common); 2709 mStandby = true; 2710 mBytesWritten = 0; 2711 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2712 keyValuePair.string()); 2713 } 2714 if (status == NO_ERROR && reconfig) { 2715 delete mAudioMixer; 2716 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2717 mAudioMixer = NULL; 2718 readOutputParameters(); 2719 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2720 for (size_t i = 0; i < mTracks.size() ; i++) { 2721 int name = getTrackName_l(); 2722 if (name < 0) break; 2723 mTracks[i]->mName = name; 2724 // limit track sample rate to 2 x new output sample rate 2725 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2726 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2727 } 2728 } 2729 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2730 } 2731 } 2732 2733 mNewParameters.removeAt(0); 2734 2735 mParamStatus = status; 2736 mParamCond.signal(); 2737 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2738 // already timed out waiting for the status and will never signal the condition. 2739 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2740 } 2741 return reconfig; 2742} 2743 2744status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2745{ 2746 const size_t SIZE = 256; 2747 char buffer[SIZE]; 2748 String8 result; 2749 2750 PlaybackThread::dumpInternals(fd, args); 2751 2752 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2753 result.append(buffer); 2754 write(fd, result.string(), result.size()); 2755 return NO_ERROR; 2756} 2757 2758uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2759{ 2760 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2761} 2762 2763uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2764{ 2765 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2766} 2767 2768// ---------------------------------------------------------------------------- 2769AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2770 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2771 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2772 // mLeftVolFloat, mRightVolFloat 2773 // mLeftVolShort, mRightVolShort 2774{ 2775} 2776 2777AudioFlinger::DirectOutputThread::~DirectOutputThread() 2778{ 2779} 2780 2781void AudioFlinger::DirectOutputThread::applyVolume() 2782{ 2783 // Do not apply volume on compressed audio 2784 if (!audio_is_linear_pcm(mFormat)) { 2785 return; 2786 } 2787 2788 // convert to signed 16 bit before volume calculation 2789 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2790 size_t count = mFrameCount * mChannelCount; 2791 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2792 int16_t *dst = mMixBuffer + count-1; 2793 while(count--) { 2794 *dst-- = (int16_t)(*src--^0x80) << 8; 2795 } 2796 } 2797 2798 size_t frameCount = mFrameCount; 2799 int16_t *out = mMixBuffer; 2800 if (rampVolume) { 2801 if (mChannelCount == 1) { 2802 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2803 int32_t vlInc = d / (int32_t)frameCount; 2804 int32_t vl = ((int32_t)mLeftVolShort << 16); 2805 do { 2806 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2807 out++; 2808 vl += vlInc; 2809 } while (--frameCount); 2810 2811 } else { 2812 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2813 int32_t vlInc = d / (int32_t)frameCount; 2814 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2815 int32_t vrInc = d / (int32_t)frameCount; 2816 int32_t vl = ((int32_t)mLeftVolShort << 16); 2817 int32_t vr = ((int32_t)mRightVolShort << 16); 2818 do { 2819 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2820 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2821 out += 2; 2822 vl += vlInc; 2823 vr += vrInc; 2824 } while (--frameCount); 2825 } 2826 } else { 2827 if (mChannelCount == 1) { 2828 do { 2829 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2830 out++; 2831 } while (--frameCount); 2832 } else { 2833 do { 2834 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2835 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2836 out += 2; 2837 } while (--frameCount); 2838 } 2839 } 2840 2841 // convert back to unsigned 8 bit after volume calculation 2842 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2843 size_t count = mFrameCount * mChannelCount; 2844 int16_t *src = mMixBuffer; 2845 uint8_t *dst = (uint8_t *)mMixBuffer; 2846 while(count--) { 2847 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2848 } 2849 } 2850 2851 mLeftVolShort = leftVol; 2852 mRightVolShort = rightVol; 2853} 2854 2855AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::threadLoop_prepareTracks_l( 2856 sp<Track>& trackToRemove 2857) 2858{ 2859 // FIXME Temporarily renamed to avoid confusion with the member "mixerStatus" 2860 mixer_state mixerStatus_ = MIXER_IDLE; 2861 2862 // find out which tracks need to be processed 2863 if (mActiveTracks.size() != 0) { 2864 sp<Track> t = mActiveTracks[0].promote(); 2865 // see FIXME in AudioFlinger.h, return MIXER_IDLE might also work 2866 if (t == 0) return MIXER_CONTINUE; 2867 //if (t == 0) continue; 2868 2869 Track* const track = t.get(); 2870 audio_track_cblk_t* cblk = track->cblk(); 2871 2872 // The first time a track is added we wait 2873 // for all its buffers to be filled before processing it 2874 if (cblk->framesReady() && track->isReady() && 2875 !track->isPaused() && !track->isTerminated()) 2876 { 2877 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2878 2879 if (track->mFillingUpStatus == Track::FS_FILLED) { 2880 track->mFillingUpStatus = Track::FS_ACTIVE; 2881 mLeftVolFloat = mRightVolFloat = 0; 2882 mLeftVolShort = mRightVolShort = 0; 2883 if (track->mState == TrackBase::RESUMING) { 2884 track->mState = TrackBase::ACTIVE; 2885 rampVolume = true; 2886 } 2887 } else if (cblk->server != 0) { 2888 // If the track is stopped before the first frame was mixed, 2889 // do not apply ramp 2890 rampVolume = true; 2891 } 2892 // compute volume for this track 2893 float left, right; 2894 if (track->isMuted() || mMasterMute || track->isPausing() || 2895 mStreamTypes[track->streamType()].mute) { 2896 left = right = 0; 2897 if (track->isPausing()) { 2898 track->setPaused(); 2899 } 2900 } else { 2901 float typeVolume = mStreamTypes[track->streamType()].volume; 2902 float v = mMasterVolume * typeVolume; 2903 uint32_t vlr = cblk->getVolumeLR(); 2904 float v_clamped = v * (vlr & 0xFFFF); 2905 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2906 left = v_clamped/MAX_GAIN; 2907 v_clamped = v * (vlr >> 16); 2908 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2909 right = v_clamped/MAX_GAIN; 2910 } 2911 2912 if (left != mLeftVolFloat || right != mRightVolFloat) { 2913 mLeftVolFloat = left; 2914 mRightVolFloat = right; 2915 2916 // If audio HAL implements volume control, 2917 // force software volume to nominal value 2918 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2919 left = 1.0f; 2920 right = 1.0f; 2921 } 2922 2923 // Convert volumes from float to 8.24 2924 uint32_t vl = (uint32_t)(left * (1 << 24)); 2925 uint32_t vr = (uint32_t)(right * (1 << 24)); 2926 2927 // Delegate volume control to effect in track effect chain if needed 2928 // only one effect chain can be present on DirectOutputThread, so if 2929 // there is one, the track is connected to it 2930 if (!mEffectChains.isEmpty()) { 2931 // Do not ramp volume if volume is controlled by effect 2932 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2933 rampVolume = false; 2934 } 2935 } 2936 2937 // Convert volumes from 8.24 to 4.12 format 2938 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2939 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2940 leftVol = (uint16_t)v_clamped; 2941 v_clamped = (vr + (1 << 11)) >> 12; 2942 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2943 rightVol = (uint16_t)v_clamped; 2944 } else { 2945 leftVol = mLeftVolShort; 2946 rightVol = mRightVolShort; 2947 rampVolume = false; 2948 } 2949 2950 // reset retry count 2951 track->mRetryCount = kMaxTrackRetriesDirect; 2952 activeTrack = t; 2953 mixerStatus_ = MIXER_TRACKS_READY; 2954 } else { 2955 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2956 if (track->isStopped()) { 2957 track->reset(); 2958 } 2959 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2960 // We have consumed all the buffers of this track. 2961 // Remove it from the list of active tracks. 2962 trackToRemove = track; 2963 } else { 2964 // No buffers for this track. Give it a few chances to 2965 // fill a buffer, then remove it from active list. 2966 if (--(track->mRetryCount) <= 0) { 2967 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2968 trackToRemove = track; 2969 } else { 2970 mixerStatus_ = MIXER_TRACKS_ENABLED; 2971 } 2972 } 2973 } 2974 } 2975 2976 // remove all the tracks that need to be... 2977 if (CC_UNLIKELY(trackToRemove != 0)) { 2978 mActiveTracks.remove(trackToRemove); 2979 if (!mEffectChains.isEmpty()) { 2980 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2981 trackToRemove->sessionId()); 2982 mEffectChains[0]->decActiveTrackCnt(); 2983 } 2984 if (trackToRemove->isTerminated()) { 2985 removeTrack_l(trackToRemove); 2986 } 2987 } 2988 2989 return mixerStatus_; 2990} 2991 2992void AudioFlinger::DirectOutputThread::threadLoop_mix() 2993{ 2994 AudioBufferProvider::Buffer buffer; 2995 size_t frameCount = mFrameCount; 2996 int8_t *curBuf = (int8_t *)mMixBuffer; 2997 // output audio to hardware 2998 while (frameCount) { 2999 buffer.frameCount = frameCount; 3000 activeTrack->getNextBuffer(&buffer); 3001 if (CC_UNLIKELY(buffer.raw == NULL)) { 3002 memset(curBuf, 0, frameCount * mFrameSize); 3003 break; 3004 } 3005 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3006 frameCount -= buffer.frameCount; 3007 curBuf += buffer.frameCount * mFrameSize; 3008 activeTrack->releaseBuffer(&buffer); 3009 } 3010 sleepTime = 0; 3011 standbyTime = systemTime() + standbyDelay; 3012} 3013 3014void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3015{ 3016 if (sleepTime == 0) { 3017 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3018 sleepTime = activeSleepTime; 3019 } else { 3020 sleepTime = idleSleepTime; 3021 } 3022 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3023 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3024 sleepTime = 0; 3025 } 3026} 3027 3028// getTrackName_l() must be called with ThreadBase::mLock held 3029int AudioFlinger::DirectOutputThread::getTrackName_l() 3030{ 3031 return 0; 3032} 3033 3034// deleteTrackName_l() must be called with ThreadBase::mLock held 3035void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3036{ 3037} 3038 3039// checkForNewParameters_l() must be called with ThreadBase::mLock held 3040bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3041{ 3042 bool reconfig = false; 3043 3044 while (!mNewParameters.isEmpty()) { 3045 status_t status = NO_ERROR; 3046 String8 keyValuePair = mNewParameters[0]; 3047 AudioParameter param = AudioParameter(keyValuePair); 3048 int value; 3049 3050 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3051 // do not accept frame count changes if tracks are open as the track buffer 3052 // size depends on frame count and correct behavior would not be garantied 3053 // if frame count is changed after track creation 3054 if (!mTracks.isEmpty()) { 3055 status = INVALID_OPERATION; 3056 } else { 3057 reconfig = true; 3058 } 3059 } 3060 if (status == NO_ERROR) { 3061 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3062 keyValuePair.string()); 3063 if (!mStandby && status == INVALID_OPERATION) { 3064 mOutput->stream->common.standby(&mOutput->stream->common); 3065 mStandby = true; 3066 mBytesWritten = 0; 3067 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3068 keyValuePair.string()); 3069 } 3070 if (status == NO_ERROR && reconfig) { 3071 readOutputParameters(); 3072 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3073 } 3074 } 3075 3076 mNewParameters.removeAt(0); 3077 3078 mParamStatus = status; 3079 mParamCond.signal(); 3080 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3081 // already timed out waiting for the status and will never signal the condition. 3082 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3083 } 3084 return reconfig; 3085} 3086 3087uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3088{ 3089 uint32_t time; 3090 if (audio_is_linear_pcm(mFormat)) { 3091 time = PlaybackThread::activeSleepTimeUs(); 3092 } else { 3093 time = 10000; 3094 } 3095 return time; 3096} 3097 3098uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3099{ 3100 uint32_t time; 3101 if (audio_is_linear_pcm(mFormat)) { 3102 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3103 } else { 3104 time = 10000; 3105 } 3106 return time; 3107} 3108 3109uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3110{ 3111 uint32_t time; 3112 if (audio_is_linear_pcm(mFormat)) { 3113 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3114 } else { 3115 time = 10000; 3116 } 3117 return time; 3118} 3119 3120 3121// ---------------------------------------------------------------------------- 3122 3123AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3124 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3125 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3126 mWaitTimeMs(UINT_MAX) 3127{ 3128 addOutputTrack(mainThread); 3129} 3130 3131AudioFlinger::DuplicatingThread::~DuplicatingThread() 3132{ 3133 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3134 mOutputTracks[i]->destroy(); 3135 } 3136} 3137 3138void AudioFlinger::DuplicatingThread::threadLoop_mix() 3139{ 3140 // mix buffers... 3141 if (outputsReady(outputTracks)) { 3142 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3143 } else { 3144 memset(mMixBuffer, 0, mixBufferSize); 3145 } 3146 sleepTime = 0; 3147 writeFrames = mFrameCount; 3148} 3149 3150void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3151{ 3152 if (sleepTime == 0) { 3153 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3154 sleepTime = activeSleepTime; 3155 } else { 3156 sleepTime = idleSleepTime; 3157 } 3158 } else if (mBytesWritten != 0) { 3159 // flush remaining overflow buffers in output tracks 3160 for (size_t i = 0; i < outputTracks.size(); i++) { 3161 if (outputTracks[i]->isActive()) { 3162 sleepTime = 0; 3163 writeFrames = 0; 3164 memset(mMixBuffer, 0, mixBufferSize); 3165 break; 3166 } 3167 } 3168 } 3169} 3170 3171void AudioFlinger::DuplicatingThread::threadLoop_write() 3172{ 3173 standbyTime = systemTime() + mStandbyTimeInNsecs; 3174 for (size_t i = 0; i < outputTracks.size(); i++) { 3175 outputTracks[i]->write(mMixBuffer, writeFrames); 3176 } 3177 mBytesWritten += mixBufferSize; 3178} 3179 3180void AudioFlinger::DuplicatingThread::threadLoop_standby() 3181{ 3182 // DuplicatingThread implements standby by stopping all tracks 3183 for (size_t i = 0; i < outputTracks.size(); i++) { 3184 outputTracks[i]->stop(); 3185 } 3186} 3187 3188void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3189{ 3190 Mutex::Autolock _l(mLock); 3191 // FIXME explain this formula 3192 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3193 OutputTrack *outputTrack = new OutputTrack(thread, 3194 this, 3195 mSampleRate, 3196 mFormat, 3197 mChannelMask, 3198 frameCount); 3199 if (outputTrack->cblk() != NULL) { 3200 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3201 mOutputTracks.add(outputTrack); 3202 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3203 updateWaitTime_l(); 3204 } 3205} 3206 3207void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3208{ 3209 Mutex::Autolock _l(mLock); 3210 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3211 if (mOutputTracks[i]->thread() == thread) { 3212 mOutputTracks[i]->destroy(); 3213 mOutputTracks.removeAt(i); 3214 updateWaitTime_l(); 3215 return; 3216 } 3217 } 3218 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3219} 3220 3221// caller must hold mLock 3222void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3223{ 3224 mWaitTimeMs = UINT_MAX; 3225 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3226 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3227 if (strong != 0) { 3228 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3229 if (waitTimeMs < mWaitTimeMs) { 3230 mWaitTimeMs = waitTimeMs; 3231 } 3232 } 3233 } 3234} 3235 3236 3237bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3238{ 3239 for (size_t i = 0; i < outputTracks.size(); i++) { 3240 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3241 if (thread == 0) { 3242 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3243 return false; 3244 } 3245 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3246 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3247 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3248 return false; 3249 } 3250 } 3251 return true; 3252} 3253 3254uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3255{ 3256 return (mWaitTimeMs * 1000) / 2; 3257} 3258 3259// ---------------------------------------------------------------------------- 3260 3261// TrackBase constructor must be called with AudioFlinger::mLock held 3262AudioFlinger::ThreadBase::TrackBase::TrackBase( 3263 ThreadBase *thread, 3264 const sp<Client>& client, 3265 uint32_t sampleRate, 3266 audio_format_t format, 3267 uint32_t channelMask, 3268 int frameCount, 3269 const sp<IMemory>& sharedBuffer, 3270 int sessionId) 3271 : RefBase(), 3272 mThread(thread), 3273 mClient(client), 3274 mCblk(NULL), 3275 // mBuffer 3276 // mBufferEnd 3277 mFrameCount(0), 3278 mState(IDLE), 3279 mFormat(format), 3280 mStepServerFailed(false), 3281 mSessionId(sessionId) 3282 // mChannelCount 3283 // mChannelMask 3284{ 3285 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3286 3287 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3288 size_t size = sizeof(audio_track_cblk_t); 3289 uint8_t channelCount = popcount(channelMask); 3290 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3291 if (sharedBuffer == 0) { 3292 size += bufferSize; 3293 } 3294 3295 if (client != NULL) { 3296 mCblkMemory = client->heap()->allocate(size); 3297 if (mCblkMemory != 0) { 3298 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3299 if (mCblk != NULL) { // construct the shared structure in-place. 3300 new(mCblk) audio_track_cblk_t(); 3301 // clear all buffers 3302 mCblk->frameCount = frameCount; 3303 mCblk->sampleRate = sampleRate; 3304 mChannelCount = channelCount; 3305 mChannelMask = channelMask; 3306 if (sharedBuffer == 0) { 3307 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3308 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3309 // Force underrun condition to avoid false underrun callback until first data is 3310 // written to buffer (other flags are cleared) 3311 mCblk->flags = CBLK_UNDERRUN_ON; 3312 } else { 3313 mBuffer = sharedBuffer->pointer(); 3314 } 3315 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3316 } 3317 } else { 3318 ALOGE("not enough memory for AudioTrack size=%u", size); 3319 client->heap()->dump("AudioTrack"); 3320 return; 3321 } 3322 } else { 3323 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3324 // construct the shared structure in-place. 3325 new(mCblk) audio_track_cblk_t(); 3326 // clear all buffers 3327 mCblk->frameCount = frameCount; 3328 mCblk->sampleRate = sampleRate; 3329 mChannelCount = channelCount; 3330 mChannelMask = channelMask; 3331 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3332 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3333 // Force underrun condition to avoid false underrun callback until first data is 3334 // written to buffer (other flags are cleared) 3335 mCblk->flags = CBLK_UNDERRUN_ON; 3336 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3337 } 3338} 3339 3340AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3341{ 3342 if (mCblk != NULL) { 3343 if (mClient == 0) { 3344 delete mCblk; 3345 } else { 3346 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3347 } 3348 } 3349 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3350 if (mClient != 0) { 3351 // Client destructor must run with AudioFlinger mutex locked 3352 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3353 // If the client's reference count drops to zero, the associated destructor 3354 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3355 // relying on the automatic clear() at end of scope. 3356 mClient.clear(); 3357 } 3358} 3359 3360// AudioBufferProvider interface 3361// getNextBuffer() = 0; 3362// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3363void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3364{ 3365 buffer->raw = NULL; 3366 mFrameCount = buffer->frameCount; 3367 (void) step(); // ignore return value of step() 3368 buffer->frameCount = 0; 3369} 3370 3371bool AudioFlinger::ThreadBase::TrackBase::step() { 3372 bool result; 3373 audio_track_cblk_t* cblk = this->cblk(); 3374 3375 result = cblk->stepServer(mFrameCount); 3376 if (!result) { 3377 ALOGV("stepServer failed acquiring cblk mutex"); 3378 mStepServerFailed = true; 3379 } 3380 return result; 3381} 3382 3383void AudioFlinger::ThreadBase::TrackBase::reset() { 3384 audio_track_cblk_t* cblk = this->cblk(); 3385 3386 cblk->user = 0; 3387 cblk->server = 0; 3388 cblk->userBase = 0; 3389 cblk->serverBase = 0; 3390 mStepServerFailed = false; 3391 ALOGV("TrackBase::reset"); 3392} 3393 3394int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3395 return (int)mCblk->sampleRate; 3396} 3397 3398void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3399 audio_track_cblk_t* cblk = this->cblk(); 3400 size_t frameSize = cblk->frameSize; 3401 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3402 int8_t *bufferEnd = bufferStart + frames * frameSize; 3403 3404 // Check validity of returned pointer in case the track control block would have been corrupted. 3405 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3406 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3407 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3408 server %d, serverBase %d, user %d, userBase %d", 3409 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3410 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3411 return NULL; 3412 } 3413 3414 return bufferStart; 3415} 3416 3417// ---------------------------------------------------------------------------- 3418 3419// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3420AudioFlinger::PlaybackThread::Track::Track( 3421 PlaybackThread *thread, 3422 const sp<Client>& client, 3423 audio_stream_type_t streamType, 3424 uint32_t sampleRate, 3425 audio_format_t format, 3426 uint32_t channelMask, 3427 int frameCount, 3428 const sp<IMemory>& sharedBuffer, 3429 int sessionId) 3430 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3431 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3432 mAuxEffectId(0), mHasVolumeController(false) 3433{ 3434 if (mCblk != NULL) { 3435 if (thread != NULL) { 3436 mName = thread->getTrackName_l(); 3437 mMainBuffer = thread->mixBuffer(); 3438 } 3439 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3440 if (mName < 0) { 3441 ALOGE("no more track names available"); 3442 } 3443 mStreamType = streamType; 3444 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3445 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3446 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3447 } 3448} 3449 3450AudioFlinger::PlaybackThread::Track::~Track() 3451{ 3452 ALOGV("PlaybackThread::Track destructor"); 3453 sp<ThreadBase> thread = mThread.promote(); 3454 if (thread != 0) { 3455 Mutex::Autolock _l(thread->mLock); 3456 mState = TERMINATED; 3457 } 3458} 3459 3460void AudioFlinger::PlaybackThread::Track::destroy() 3461{ 3462 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3463 // by removing it from mTracks vector, so there is a risk that this Tracks's 3464 // destructor is called. As the destructor needs to lock mLock, 3465 // we must acquire a strong reference on this Track before locking mLock 3466 // here so that the destructor is called only when exiting this function. 3467 // On the other hand, as long as Track::destroy() is only called by 3468 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3469 // this Track with its member mTrack. 3470 sp<Track> keep(this); 3471 { // scope for mLock 3472 sp<ThreadBase> thread = mThread.promote(); 3473 if (thread != 0) { 3474 if (!isOutputTrack()) { 3475 if (mState == ACTIVE || mState == RESUMING) { 3476 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3477 3478 // to track the speaker usage 3479 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3480 } 3481 AudioSystem::releaseOutput(thread->id()); 3482 } 3483 Mutex::Autolock _l(thread->mLock); 3484 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3485 playbackThread->destroyTrack_l(this); 3486 } 3487 } 3488} 3489 3490void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3491{ 3492 uint32_t vlr = mCblk->getVolumeLR(); 3493 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", 3494 mName - AudioMixer::TRACK0, 3495 (mClient == 0) ? getpid_cached : mClient->pid(), 3496 mStreamType, 3497 mFormat, 3498 mChannelMask, 3499 mSessionId, 3500 mFrameCount, 3501 mState, 3502 mMute, 3503 mFillingUpStatus, 3504 mCblk->sampleRate, 3505 vlr & 0xFFFF, 3506 vlr >> 16, 3507 mCblk->server, 3508 mCblk->user, 3509 (int)mMainBuffer, 3510 (int)mAuxBuffer); 3511} 3512 3513// AudioBufferProvider interface 3514status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3515 AudioBufferProvider::Buffer* buffer, int64_t pts) 3516{ 3517 audio_track_cblk_t* cblk = this->cblk(); 3518 uint32_t framesReady; 3519 uint32_t framesReq = buffer->frameCount; 3520 3521 // Check if last stepServer failed, try to step now 3522 if (mStepServerFailed) { 3523 if (!step()) goto getNextBuffer_exit; 3524 ALOGV("stepServer recovered"); 3525 mStepServerFailed = false; 3526 } 3527 3528 framesReady = cblk->framesReady(); 3529 3530 if (CC_LIKELY(framesReady)) { 3531 uint32_t s = cblk->server; 3532 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3533 3534 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3535 if (framesReq > framesReady) { 3536 framesReq = framesReady; 3537 } 3538 if (s + framesReq > bufferEnd) { 3539 framesReq = bufferEnd - s; 3540 } 3541 3542 buffer->raw = getBuffer(s, framesReq); 3543 if (buffer->raw == NULL) goto getNextBuffer_exit; 3544 3545 buffer->frameCount = framesReq; 3546 return NO_ERROR; 3547 } 3548 3549getNextBuffer_exit: 3550 buffer->raw = NULL; 3551 buffer->frameCount = 0; 3552 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3553 return NOT_ENOUGH_DATA; 3554} 3555 3556uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3557 return mCblk->framesReady(); 3558} 3559 3560bool AudioFlinger::PlaybackThread::Track::isReady() const { 3561 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3562 3563 if (framesReady() >= mCblk->frameCount || 3564 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3565 mFillingUpStatus = FS_FILLED; 3566 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3567 return true; 3568 } 3569 return false; 3570} 3571 3572status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3573{ 3574 status_t status = NO_ERROR; 3575 ALOGV("start(%d), calling pid %d session %d tid %d", 3576 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3577 sp<ThreadBase> thread = mThread.promote(); 3578 if (thread != 0) { 3579 Mutex::Autolock _l(thread->mLock); 3580 track_state state = mState; 3581 // here the track could be either new, or restarted 3582 // in both cases "unstop" the track 3583 if (mState == PAUSED) { 3584 mState = TrackBase::RESUMING; 3585 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3586 } else { 3587 mState = TrackBase::ACTIVE; 3588 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3589 } 3590 3591 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3592 thread->mLock.unlock(); 3593 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3594 thread->mLock.lock(); 3595 3596 // to track the speaker usage 3597 if (status == NO_ERROR) { 3598 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3599 } 3600 } 3601 if (status == NO_ERROR) { 3602 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3603 playbackThread->addTrack_l(this); 3604 } else { 3605 mState = state; 3606 } 3607 } else { 3608 status = BAD_VALUE; 3609 } 3610 return status; 3611} 3612 3613void AudioFlinger::PlaybackThread::Track::stop() 3614{ 3615 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3616 sp<ThreadBase> thread = mThread.promote(); 3617 if (thread != 0) { 3618 Mutex::Autolock _l(thread->mLock); 3619 track_state state = mState; 3620 if (mState > STOPPED) { 3621 mState = STOPPED; 3622 // If the track is not active (PAUSED and buffers full), flush buffers 3623 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3624 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3625 reset(); 3626 } 3627 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3628 } 3629 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3630 thread->mLock.unlock(); 3631 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3632 thread->mLock.lock(); 3633 3634 // to track the speaker usage 3635 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3636 } 3637 } 3638} 3639 3640void AudioFlinger::PlaybackThread::Track::pause() 3641{ 3642 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3643 sp<ThreadBase> thread = mThread.promote(); 3644 if (thread != 0) { 3645 Mutex::Autolock _l(thread->mLock); 3646 if (mState == ACTIVE || mState == RESUMING) { 3647 mState = PAUSING; 3648 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3649 if (!isOutputTrack()) { 3650 thread->mLock.unlock(); 3651 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3652 thread->mLock.lock(); 3653 3654 // to track the speaker usage 3655 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3656 } 3657 } 3658 } 3659} 3660 3661void AudioFlinger::PlaybackThread::Track::flush() 3662{ 3663 ALOGV("flush(%d)", mName); 3664 sp<ThreadBase> thread = mThread.promote(); 3665 if (thread != 0) { 3666 Mutex::Autolock _l(thread->mLock); 3667 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3668 return; 3669 } 3670 // No point remaining in PAUSED state after a flush => go to 3671 // STOPPED state 3672 mState = STOPPED; 3673 3674 // do not reset the track if it is still in the process of being stopped or paused. 3675 // this will be done by prepareTracks_l() when the track is stopped. 3676 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3677 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3678 reset(); 3679 } 3680 } 3681} 3682 3683void AudioFlinger::PlaybackThread::Track::reset() 3684{ 3685 // Do not reset twice to avoid discarding data written just after a flush and before 3686 // the audioflinger thread detects the track is stopped. 3687 if (!mResetDone) { 3688 TrackBase::reset(); 3689 // Force underrun condition to avoid false underrun callback until first data is 3690 // written to buffer 3691 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3692 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3693 mFillingUpStatus = FS_FILLING; 3694 mResetDone = true; 3695 } 3696} 3697 3698void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3699{ 3700 mMute = muted; 3701} 3702 3703status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3704{ 3705 status_t status = DEAD_OBJECT; 3706 sp<ThreadBase> thread = mThread.promote(); 3707 if (thread != 0) { 3708 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3709 status = playbackThread->attachAuxEffect(this, EffectId); 3710 } 3711 return status; 3712} 3713 3714void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3715{ 3716 mAuxEffectId = EffectId; 3717 mAuxBuffer = buffer; 3718} 3719 3720// timed audio tracks 3721 3722sp<AudioFlinger::PlaybackThread::TimedTrack> 3723AudioFlinger::PlaybackThread::TimedTrack::create( 3724 PlaybackThread *thread, 3725 const sp<Client>& client, 3726 audio_stream_type_t streamType, 3727 uint32_t sampleRate, 3728 audio_format_t format, 3729 uint32_t channelMask, 3730 int frameCount, 3731 const sp<IMemory>& sharedBuffer, 3732 int sessionId) { 3733 if (!client->reserveTimedTrack()) 3734 return NULL; 3735 3736 sp<TimedTrack> track = new TimedTrack( 3737 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3738 sharedBuffer, sessionId); 3739 3740 if (track == NULL) { 3741 client->releaseTimedTrack(); 3742 return NULL; 3743 } 3744 3745 return track; 3746} 3747 3748AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3749 PlaybackThread *thread, 3750 const sp<Client>& client, 3751 audio_stream_type_t streamType, 3752 uint32_t sampleRate, 3753 audio_format_t format, 3754 uint32_t channelMask, 3755 int frameCount, 3756 const sp<IMemory>& sharedBuffer, 3757 int sessionId) 3758 : Track(thread, client, streamType, sampleRate, format, channelMask, 3759 frameCount, sharedBuffer, sessionId), 3760 mTimedSilenceBuffer(NULL), 3761 mTimedSilenceBufferSize(0), 3762 mTimedAudioOutputOnTime(false), 3763 mMediaTimeTransformValid(false) 3764{ 3765 LocalClock lc; 3766 mLocalTimeFreq = lc.getLocalFreq(); 3767 3768 mLocalTimeToSampleTransform.a_zero = 0; 3769 mLocalTimeToSampleTransform.b_zero = 0; 3770 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3771 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3772 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3773 &mLocalTimeToSampleTransform.a_to_b_denom); 3774} 3775 3776AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3777 mClient->releaseTimedTrack(); 3778 delete [] mTimedSilenceBuffer; 3779} 3780 3781status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3782 size_t size, sp<IMemory>* buffer) { 3783 3784 Mutex::Autolock _l(mTimedBufferQueueLock); 3785 3786 trimTimedBufferQueue_l(); 3787 3788 // lazily initialize the shared memory heap for timed buffers 3789 if (mTimedMemoryDealer == NULL) { 3790 const int kTimedBufferHeapSize = 512 << 10; 3791 3792 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3793 "AudioFlingerTimed"); 3794 if (mTimedMemoryDealer == NULL) 3795 return NO_MEMORY; 3796 } 3797 3798 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3799 if (newBuffer == NULL) { 3800 newBuffer = mTimedMemoryDealer->allocate(size); 3801 if (newBuffer == NULL) 3802 return NO_MEMORY; 3803 } 3804 3805 *buffer = newBuffer; 3806 return NO_ERROR; 3807} 3808 3809// caller must hold mTimedBufferQueueLock 3810void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3811 int64_t mediaTimeNow; 3812 { 3813 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3814 if (!mMediaTimeTransformValid) 3815 return; 3816 3817 int64_t targetTimeNow; 3818 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3819 ? mCCHelper.getCommonTime(&targetTimeNow) 3820 : mCCHelper.getLocalTime(&targetTimeNow); 3821 3822 if (OK != res) 3823 return; 3824 3825 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3826 &mediaTimeNow)) { 3827 return; 3828 } 3829 } 3830 3831 size_t trimIndex; 3832 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3833 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3834 break; 3835 } 3836 3837 if (trimIndex) { 3838 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3839 } 3840} 3841 3842status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3843 const sp<IMemory>& buffer, int64_t pts) { 3844 3845 { 3846 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3847 if (!mMediaTimeTransformValid) 3848 return INVALID_OPERATION; 3849 } 3850 3851 Mutex::Autolock _l(mTimedBufferQueueLock); 3852 3853 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3854 3855 return NO_ERROR; 3856} 3857 3858status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3859 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3860 3861 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3862 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3863 target); 3864 3865 if (!(target == TimedAudioTrack::LOCAL_TIME || 3866 target == TimedAudioTrack::COMMON_TIME)) { 3867 return BAD_VALUE; 3868 } 3869 3870 Mutex::Autolock lock(mMediaTimeTransformLock); 3871 mMediaTimeTransform = xform; 3872 mMediaTimeTransformTarget = target; 3873 mMediaTimeTransformValid = true; 3874 3875 return NO_ERROR; 3876} 3877 3878#define min(a, b) ((a) < (b) ? (a) : (b)) 3879 3880// implementation of getNextBuffer for tracks whose buffers have timestamps 3881status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3882 AudioBufferProvider::Buffer* buffer, int64_t pts) 3883{ 3884 if (pts == AudioBufferProvider::kInvalidPTS) { 3885 buffer->raw = 0; 3886 buffer->frameCount = 0; 3887 return INVALID_OPERATION; 3888 } 3889 3890 Mutex::Autolock _l(mTimedBufferQueueLock); 3891 3892 while (true) { 3893 3894 // if we have no timed buffers, then fail 3895 if (mTimedBufferQueue.isEmpty()) { 3896 buffer->raw = 0; 3897 buffer->frameCount = 0; 3898 return NOT_ENOUGH_DATA; 3899 } 3900 3901 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3902 3903 // calculate the PTS of the head of the timed buffer queue expressed in 3904 // local time 3905 int64_t headLocalPTS; 3906 { 3907 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3908 3909 assert(mMediaTimeTransformValid); 3910 3911 if (mMediaTimeTransform.a_to_b_denom == 0) { 3912 // the transform represents a pause, so yield silence 3913 timedYieldSilence(buffer->frameCount, buffer); 3914 return NO_ERROR; 3915 } 3916 3917 int64_t transformedPTS; 3918 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3919 &transformedPTS)) { 3920 // the transform failed. this shouldn't happen, but if it does 3921 // then just drop this buffer 3922 ALOGW("timedGetNextBuffer transform failed"); 3923 buffer->raw = 0; 3924 buffer->frameCount = 0; 3925 mTimedBufferQueue.removeAt(0); 3926 return NO_ERROR; 3927 } 3928 3929 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3930 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3931 &headLocalPTS)) { 3932 buffer->raw = 0; 3933 buffer->frameCount = 0; 3934 return INVALID_OPERATION; 3935 } 3936 } else { 3937 headLocalPTS = transformedPTS; 3938 } 3939 } 3940 3941 // adjust the head buffer's PTS to reflect the portion of the head buffer 3942 // that has already been consumed 3943 int64_t effectivePTS = headLocalPTS + 3944 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3945 3946 // Calculate the delta in samples between the head of the input buffer 3947 // queue and the start of the next output buffer that will be written. 3948 // If the transformation fails because of over or underflow, it means 3949 // that the sample's position in the output stream is so far out of 3950 // whack that it should just be dropped. 3951 int64_t sampleDelta; 3952 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3953 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3954 mTimedBufferQueue.removeAt(0); 3955 continue; 3956 } 3957 if (!mLocalTimeToSampleTransform.doForwardTransform( 3958 (effectivePTS - pts) << 32, &sampleDelta)) { 3959 ALOGV("*** too late during sample rate transform: dropped buffer"); 3960 mTimedBufferQueue.removeAt(0); 3961 continue; 3962 } 3963 3964 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3965 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3966 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3967 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3968 3969 // if the delta between the ideal placement for the next input sample and 3970 // the current output position is within this threshold, then we will 3971 // concatenate the next input samples to the previous output 3972 const int64_t kSampleContinuityThreshold = 3973 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3974 3975 // if this is the first buffer of audio that we're emitting from this track 3976 // then it should be almost exactly on time. 3977 const int64_t kSampleStartupThreshold = 1LL << 32; 3978 3979 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3980 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3981 // the next input is close enough to being on time, so concatenate it 3982 // with the last output 3983 timedYieldSamples(buffer); 3984 3985 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3986 return NO_ERROR; 3987 } else if (sampleDelta > 0) { 3988 // the gap between the current output position and the proper start of 3989 // the next input sample is too big, so fill it with silence 3990 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3991 3992 timedYieldSilence(framesUntilNextInput, buffer); 3993 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3994 return NO_ERROR; 3995 } else { 3996 // the next input sample is late 3997 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3998 size_t onTimeSamplePosition = 3999 head.position() + lateFrames * mCblk->frameSize; 4000 4001 if (onTimeSamplePosition > head.buffer()->size()) { 4002 // all the remaining samples in the head are too late, so 4003 // drop it and move on 4004 ALOGV("*** too late: dropped buffer"); 4005 mTimedBufferQueue.removeAt(0); 4006 continue; 4007 } else { 4008 // skip over the late samples 4009 head.setPosition(onTimeSamplePosition); 4010 4011 // yield the available samples 4012 timedYieldSamples(buffer); 4013 4014 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4015 return NO_ERROR; 4016 } 4017 } 4018 } 4019} 4020 4021// Yield samples from the timed buffer queue head up to the given output 4022// buffer's capacity. 4023// 4024// Caller must hold mTimedBufferQueueLock 4025void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4026 AudioBufferProvider::Buffer* buffer) { 4027 4028 const TimedBuffer& head = mTimedBufferQueue[0]; 4029 4030 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4031 head.position()); 4032 4033 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4034 mCblk->frameSize); 4035 size_t framesRequested = buffer->frameCount; 4036 buffer->frameCount = min(framesLeftInHead, framesRequested); 4037 4038 mTimedAudioOutputOnTime = true; 4039} 4040 4041// Yield samples of silence up to the given output buffer's capacity 4042// 4043// Caller must hold mTimedBufferQueueLock 4044void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4045 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4046 4047 // lazily allocate a buffer filled with silence 4048 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4049 delete [] mTimedSilenceBuffer; 4050 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4051 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4052 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4053 } 4054 4055 buffer->raw = mTimedSilenceBuffer; 4056 size_t framesRequested = buffer->frameCount; 4057 buffer->frameCount = min(numFrames, framesRequested); 4058 4059 mTimedAudioOutputOnTime = false; 4060} 4061 4062// AudioBufferProvider interface 4063void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4064 AudioBufferProvider::Buffer* buffer) { 4065 4066 Mutex::Autolock _l(mTimedBufferQueueLock); 4067 4068 // If the buffer which was just released is part of the buffer at the head 4069 // of the queue, be sure to update the amt of the buffer which has been 4070 // consumed. If the buffer being returned is not part of the head of the 4071 // queue, its either because the buffer is part of the silence buffer, or 4072 // because the head of the timed queue was trimmed after the mixer called 4073 // getNextBuffer but before the mixer called releaseBuffer. 4074 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4075 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4076 4077 void* start = head.buffer()->pointer(); 4078 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4079 4080 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4081 head.setPosition(head.position() + 4082 (buffer->frameCount * mCblk->frameSize)); 4083 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4084 mTimedBufferQueue.removeAt(0); 4085 } 4086 } 4087 } 4088 4089 buffer->raw = 0; 4090 buffer->frameCount = 0; 4091} 4092 4093uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4094 Mutex::Autolock _l(mTimedBufferQueueLock); 4095 4096 uint32_t frames = 0; 4097 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4098 const TimedBuffer& tb = mTimedBufferQueue[i]; 4099 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4100 } 4101 4102 return frames; 4103} 4104 4105AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4106 : mPTS(0), mPosition(0) {} 4107 4108AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4109 const sp<IMemory>& buffer, int64_t pts) 4110 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4111 4112// ---------------------------------------------------------------------------- 4113 4114// RecordTrack constructor must be called with AudioFlinger::mLock held 4115AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4116 RecordThread *thread, 4117 const sp<Client>& client, 4118 uint32_t sampleRate, 4119 audio_format_t format, 4120 uint32_t channelMask, 4121 int frameCount, 4122 int sessionId) 4123 : TrackBase(thread, client, sampleRate, format, 4124 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4125 mOverflow(false) 4126{ 4127 if (mCblk != NULL) { 4128 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4129 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4130 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4131 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4132 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4133 } else { 4134 mCblk->frameSize = sizeof(int8_t); 4135 } 4136 } 4137} 4138 4139AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4140{ 4141 sp<ThreadBase> thread = mThread.promote(); 4142 if (thread != 0) { 4143 AudioSystem::releaseInput(thread->id()); 4144 } 4145} 4146 4147// AudioBufferProvider interface 4148status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4149{ 4150 audio_track_cblk_t* cblk = this->cblk(); 4151 uint32_t framesAvail; 4152 uint32_t framesReq = buffer->frameCount; 4153 4154 // Check if last stepServer failed, try to step now 4155 if (mStepServerFailed) { 4156 if (!step()) goto getNextBuffer_exit; 4157 ALOGV("stepServer recovered"); 4158 mStepServerFailed = false; 4159 } 4160 4161 framesAvail = cblk->framesAvailable_l(); 4162 4163 if (CC_LIKELY(framesAvail)) { 4164 uint32_t s = cblk->server; 4165 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4166 4167 if (framesReq > framesAvail) { 4168 framesReq = framesAvail; 4169 } 4170 if (s + framesReq > bufferEnd) { 4171 framesReq = bufferEnd - s; 4172 } 4173 4174 buffer->raw = getBuffer(s, framesReq); 4175 if (buffer->raw == NULL) goto getNextBuffer_exit; 4176 4177 buffer->frameCount = framesReq; 4178 return NO_ERROR; 4179 } 4180 4181getNextBuffer_exit: 4182 buffer->raw = NULL; 4183 buffer->frameCount = 0; 4184 return NOT_ENOUGH_DATA; 4185} 4186 4187status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4188{ 4189 sp<ThreadBase> thread = mThread.promote(); 4190 if (thread != 0) { 4191 RecordThread *recordThread = (RecordThread *)thread.get(); 4192 return recordThread->start(this, tid); 4193 } else { 4194 return BAD_VALUE; 4195 } 4196} 4197 4198void AudioFlinger::RecordThread::RecordTrack::stop() 4199{ 4200 sp<ThreadBase> thread = mThread.promote(); 4201 if (thread != 0) { 4202 RecordThread *recordThread = (RecordThread *)thread.get(); 4203 recordThread->stop(this); 4204 TrackBase::reset(); 4205 // Force overerrun condition to avoid false overrun callback until first data is 4206 // read from buffer 4207 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4208 } 4209} 4210 4211void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4212{ 4213 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4214 (mClient == 0) ? getpid_cached : mClient->pid(), 4215 mFormat, 4216 mChannelMask, 4217 mSessionId, 4218 mFrameCount, 4219 mState, 4220 mCblk->sampleRate, 4221 mCblk->server, 4222 mCblk->user); 4223} 4224 4225 4226// ---------------------------------------------------------------------------- 4227 4228AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4229 PlaybackThread *playbackThread, 4230 DuplicatingThread *sourceThread, 4231 uint32_t sampleRate, 4232 audio_format_t format, 4233 uint32_t channelMask, 4234 int frameCount) 4235 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4236 mActive(false), mSourceThread(sourceThread) 4237{ 4238 4239 if (mCblk != NULL) { 4240 mCblk->flags |= CBLK_DIRECTION_OUT; 4241 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4242 mOutBuffer.frameCount = 0; 4243 playbackThread->mTracks.add(this); 4244 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4245 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4246 mCblk, mBuffer, mCblk->buffers, 4247 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4248 } else { 4249 ALOGW("Error creating output track on thread %p", playbackThread); 4250 } 4251} 4252 4253AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4254{ 4255 clearBufferQueue(); 4256} 4257 4258status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4259{ 4260 status_t status = Track::start(tid); 4261 if (status != NO_ERROR) { 4262 return status; 4263 } 4264 4265 mActive = true; 4266 mRetryCount = 127; 4267 return status; 4268} 4269 4270void AudioFlinger::PlaybackThread::OutputTrack::stop() 4271{ 4272 Track::stop(); 4273 clearBufferQueue(); 4274 mOutBuffer.frameCount = 0; 4275 mActive = false; 4276} 4277 4278bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4279{ 4280 Buffer *pInBuffer; 4281 Buffer inBuffer; 4282 uint32_t channelCount = mChannelCount; 4283 bool outputBufferFull = false; 4284 inBuffer.frameCount = frames; 4285 inBuffer.i16 = data; 4286 4287 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4288 4289 if (!mActive && frames != 0) { 4290 start(0); 4291 sp<ThreadBase> thread = mThread.promote(); 4292 if (thread != 0) { 4293 MixerThread *mixerThread = (MixerThread *)thread.get(); 4294 if (mCblk->frameCount > frames){ 4295 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4296 uint32_t startFrames = (mCblk->frameCount - frames); 4297 pInBuffer = new Buffer; 4298 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4299 pInBuffer->frameCount = startFrames; 4300 pInBuffer->i16 = pInBuffer->mBuffer; 4301 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4302 mBufferQueue.add(pInBuffer); 4303 } else { 4304 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4305 } 4306 } 4307 } 4308 } 4309 4310 while (waitTimeLeftMs) { 4311 // First write pending buffers, then new data 4312 if (mBufferQueue.size()) { 4313 pInBuffer = mBufferQueue.itemAt(0); 4314 } else { 4315 pInBuffer = &inBuffer; 4316 } 4317 4318 if (pInBuffer->frameCount == 0) { 4319 break; 4320 } 4321 4322 if (mOutBuffer.frameCount == 0) { 4323 mOutBuffer.frameCount = pInBuffer->frameCount; 4324 nsecs_t startTime = systemTime(); 4325 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4326 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4327 outputBufferFull = true; 4328 break; 4329 } 4330 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4331 if (waitTimeLeftMs >= waitTimeMs) { 4332 waitTimeLeftMs -= waitTimeMs; 4333 } else { 4334 waitTimeLeftMs = 0; 4335 } 4336 } 4337 4338 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4339 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4340 mCblk->stepUser(outFrames); 4341 pInBuffer->frameCount -= outFrames; 4342 pInBuffer->i16 += outFrames * channelCount; 4343 mOutBuffer.frameCount -= outFrames; 4344 mOutBuffer.i16 += outFrames * channelCount; 4345 4346 if (pInBuffer->frameCount == 0) { 4347 if (mBufferQueue.size()) { 4348 mBufferQueue.removeAt(0); 4349 delete [] pInBuffer->mBuffer; 4350 delete pInBuffer; 4351 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4352 } else { 4353 break; 4354 } 4355 } 4356 } 4357 4358 // If we could not write all frames, allocate a buffer and queue it for next time. 4359 if (inBuffer.frameCount) { 4360 sp<ThreadBase> thread = mThread.promote(); 4361 if (thread != 0 && !thread->standby()) { 4362 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4363 pInBuffer = new Buffer; 4364 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4365 pInBuffer->frameCount = inBuffer.frameCount; 4366 pInBuffer->i16 = pInBuffer->mBuffer; 4367 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4368 mBufferQueue.add(pInBuffer); 4369 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4370 } else { 4371 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4372 } 4373 } 4374 } 4375 4376 // Calling write() with a 0 length buffer, means that no more data will be written: 4377 // If no more buffers are pending, fill output track buffer to make sure it is started 4378 // by output mixer. 4379 if (frames == 0 && mBufferQueue.size() == 0) { 4380 if (mCblk->user < mCblk->frameCount) { 4381 frames = mCblk->frameCount - mCblk->user; 4382 pInBuffer = new Buffer; 4383 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4384 pInBuffer->frameCount = frames; 4385 pInBuffer->i16 = pInBuffer->mBuffer; 4386 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4387 mBufferQueue.add(pInBuffer); 4388 } else if (mActive) { 4389 stop(); 4390 } 4391 } 4392 4393 return outputBufferFull; 4394} 4395 4396status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4397{ 4398 int active; 4399 status_t result; 4400 audio_track_cblk_t* cblk = mCblk; 4401 uint32_t framesReq = buffer->frameCount; 4402 4403// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4404 buffer->frameCount = 0; 4405 4406 uint32_t framesAvail = cblk->framesAvailable(); 4407 4408 4409 if (framesAvail == 0) { 4410 Mutex::Autolock _l(cblk->lock); 4411 goto start_loop_here; 4412 while (framesAvail == 0) { 4413 active = mActive; 4414 if (CC_UNLIKELY(!active)) { 4415 ALOGV("Not active and NO_MORE_BUFFERS"); 4416 return NO_MORE_BUFFERS; 4417 } 4418 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4419 if (result != NO_ERROR) { 4420 return NO_MORE_BUFFERS; 4421 } 4422 // read the server count again 4423 start_loop_here: 4424 framesAvail = cblk->framesAvailable_l(); 4425 } 4426 } 4427 4428// if (framesAvail < framesReq) { 4429// return NO_MORE_BUFFERS; 4430// } 4431 4432 if (framesReq > framesAvail) { 4433 framesReq = framesAvail; 4434 } 4435 4436 uint32_t u = cblk->user; 4437 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4438 4439 if (u + framesReq > bufferEnd) { 4440 framesReq = bufferEnd - u; 4441 } 4442 4443 buffer->frameCount = framesReq; 4444 buffer->raw = (void *)cblk->buffer(u); 4445 return NO_ERROR; 4446} 4447 4448 4449void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4450{ 4451 size_t size = mBufferQueue.size(); 4452 4453 for (size_t i = 0; i < size; i++) { 4454 Buffer *pBuffer = mBufferQueue.itemAt(i); 4455 delete [] pBuffer->mBuffer; 4456 delete pBuffer; 4457 } 4458 mBufferQueue.clear(); 4459} 4460 4461// ---------------------------------------------------------------------------- 4462 4463AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4464 : RefBase(), 4465 mAudioFlinger(audioFlinger), 4466 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4467 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4468 mPid(pid), 4469 mTimedTrackCount(0) 4470{ 4471 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4472} 4473 4474// Client destructor must be called with AudioFlinger::mLock held 4475AudioFlinger::Client::~Client() 4476{ 4477 mAudioFlinger->removeClient_l(mPid); 4478} 4479 4480sp<MemoryDealer> AudioFlinger::Client::heap() const 4481{ 4482 return mMemoryDealer; 4483} 4484 4485// Reserve one of the limited slots for a timed audio track associated 4486// with this client 4487bool AudioFlinger::Client::reserveTimedTrack() 4488{ 4489 const int kMaxTimedTracksPerClient = 4; 4490 4491 Mutex::Autolock _l(mTimedTrackLock); 4492 4493 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4494 ALOGW("can not create timed track - pid %d has exceeded the limit", 4495 mPid); 4496 return false; 4497 } 4498 4499 mTimedTrackCount++; 4500 return true; 4501} 4502 4503// Release a slot for a timed audio track 4504void AudioFlinger::Client::releaseTimedTrack() 4505{ 4506 Mutex::Autolock _l(mTimedTrackLock); 4507 mTimedTrackCount--; 4508} 4509 4510// ---------------------------------------------------------------------------- 4511 4512AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4513 const sp<IAudioFlingerClient>& client, 4514 pid_t pid) 4515 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4516{ 4517} 4518 4519AudioFlinger::NotificationClient::~NotificationClient() 4520{ 4521} 4522 4523void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4524{ 4525 sp<NotificationClient> keep(this); 4526 mAudioFlinger->removeNotificationClient(mPid); 4527} 4528 4529// ---------------------------------------------------------------------------- 4530 4531AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4532 : BnAudioTrack(), 4533 mTrack(track) 4534{ 4535} 4536 4537AudioFlinger::TrackHandle::~TrackHandle() { 4538 // just stop the track on deletion, associated resources 4539 // will be freed from the main thread once all pending buffers have 4540 // been played. Unless it's not in the active track list, in which 4541 // case we free everything now... 4542 mTrack->destroy(); 4543} 4544 4545sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4546 return mTrack->getCblk(); 4547} 4548 4549status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4550 return mTrack->start(tid); 4551} 4552 4553void AudioFlinger::TrackHandle::stop() { 4554 mTrack->stop(); 4555} 4556 4557void AudioFlinger::TrackHandle::flush() { 4558 mTrack->flush(); 4559} 4560 4561void AudioFlinger::TrackHandle::mute(bool e) { 4562 mTrack->mute(e); 4563} 4564 4565void AudioFlinger::TrackHandle::pause() { 4566 mTrack->pause(); 4567} 4568 4569status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4570{ 4571 return mTrack->attachAuxEffect(EffectId); 4572} 4573 4574status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4575 sp<IMemory>* buffer) { 4576 if (!mTrack->isTimedTrack()) 4577 return INVALID_OPERATION; 4578 4579 PlaybackThread::TimedTrack* tt = 4580 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4581 return tt->allocateTimedBuffer(size, buffer); 4582} 4583 4584status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4585 int64_t pts) { 4586 if (!mTrack->isTimedTrack()) 4587 return INVALID_OPERATION; 4588 4589 PlaybackThread::TimedTrack* tt = 4590 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4591 return tt->queueTimedBuffer(buffer, pts); 4592} 4593 4594status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4595 const LinearTransform& xform, int target) { 4596 4597 if (!mTrack->isTimedTrack()) 4598 return INVALID_OPERATION; 4599 4600 PlaybackThread::TimedTrack* tt = 4601 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4602 return tt->setMediaTimeTransform( 4603 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4604} 4605 4606status_t AudioFlinger::TrackHandle::onTransact( 4607 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4608{ 4609 return BnAudioTrack::onTransact(code, data, reply, flags); 4610} 4611 4612// ---------------------------------------------------------------------------- 4613 4614sp<IAudioRecord> AudioFlinger::openRecord( 4615 pid_t pid, 4616 audio_io_handle_t input, 4617 uint32_t sampleRate, 4618 audio_format_t format, 4619 uint32_t channelMask, 4620 int frameCount, 4621 // FIXME dead, remove from IAudioFlinger 4622 uint32_t flags, 4623 int *sessionId, 4624 status_t *status) 4625{ 4626 sp<RecordThread::RecordTrack> recordTrack; 4627 sp<RecordHandle> recordHandle; 4628 sp<Client> client; 4629 status_t lStatus; 4630 RecordThread *thread; 4631 size_t inFrameCount; 4632 int lSessionId; 4633 4634 // check calling permissions 4635 if (!recordingAllowed()) { 4636 lStatus = PERMISSION_DENIED; 4637 goto Exit; 4638 } 4639 4640 // add client to list 4641 { // scope for mLock 4642 Mutex::Autolock _l(mLock); 4643 thread = checkRecordThread_l(input); 4644 if (thread == NULL) { 4645 lStatus = BAD_VALUE; 4646 goto Exit; 4647 } 4648 4649 client = registerPid_l(pid); 4650 4651 // If no audio session id is provided, create one here 4652 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4653 lSessionId = *sessionId; 4654 } else { 4655 lSessionId = nextUniqueId(); 4656 if (sessionId != NULL) { 4657 *sessionId = lSessionId; 4658 } 4659 } 4660 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4661 recordTrack = thread->createRecordTrack_l(client, 4662 sampleRate, 4663 format, 4664 channelMask, 4665 frameCount, 4666 lSessionId, 4667 &lStatus); 4668 } 4669 if (lStatus != NO_ERROR) { 4670 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4671 // destructor is called by the TrackBase destructor with mLock held 4672 client.clear(); 4673 recordTrack.clear(); 4674 goto Exit; 4675 } 4676 4677 // return to handle to client 4678 recordHandle = new RecordHandle(recordTrack); 4679 lStatus = NO_ERROR; 4680 4681Exit: 4682 if (status) { 4683 *status = lStatus; 4684 } 4685 return recordHandle; 4686} 4687 4688// ---------------------------------------------------------------------------- 4689 4690AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4691 : BnAudioRecord(), 4692 mRecordTrack(recordTrack) 4693{ 4694} 4695 4696AudioFlinger::RecordHandle::~RecordHandle() { 4697 stop(); 4698} 4699 4700sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4701 return mRecordTrack->getCblk(); 4702} 4703 4704status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4705 ALOGV("RecordHandle::start()"); 4706 return mRecordTrack->start(tid); 4707} 4708 4709void AudioFlinger::RecordHandle::stop() { 4710 ALOGV("RecordHandle::stop()"); 4711 mRecordTrack->stop(); 4712} 4713 4714status_t AudioFlinger::RecordHandle::onTransact( 4715 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4716{ 4717 return BnAudioRecord::onTransact(code, data, reply, flags); 4718} 4719 4720// ---------------------------------------------------------------------------- 4721 4722AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4723 AudioStreamIn *input, 4724 uint32_t sampleRate, 4725 uint32_t channels, 4726 audio_io_handle_t id, 4727 uint32_t device) : 4728 ThreadBase(audioFlinger, id, device, RECORD), 4729 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4730 // mRsmpInIndex and mInputBytes set by readInputParameters() 4731 mReqChannelCount(popcount(channels)), 4732 mReqSampleRate(sampleRate) 4733 // mBytesRead is only meaningful while active, and so is cleared in start() 4734 // (but might be better to also clear here for dump?) 4735{ 4736 snprintf(mName, kNameLength, "AudioIn_%X", id); 4737 4738 readInputParameters(); 4739} 4740 4741 4742AudioFlinger::RecordThread::~RecordThread() 4743{ 4744 delete[] mRsmpInBuffer; 4745 delete mResampler; 4746 delete[] mRsmpOutBuffer; 4747} 4748 4749void AudioFlinger::RecordThread::onFirstRef() 4750{ 4751 run(mName, PRIORITY_URGENT_AUDIO); 4752} 4753 4754status_t AudioFlinger::RecordThread::readyToRun() 4755{ 4756 status_t status = initCheck(); 4757 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4758 return status; 4759} 4760 4761bool AudioFlinger::RecordThread::threadLoop() 4762{ 4763 AudioBufferProvider::Buffer buffer; 4764 sp<RecordTrack> activeTrack; 4765 Vector< sp<EffectChain> > effectChains; 4766 4767 nsecs_t lastWarning = 0; 4768 4769 acquireWakeLock(); 4770 4771 // start recording 4772 while (!exitPending()) { 4773 4774 processConfigEvents(); 4775 4776 { // scope for mLock 4777 Mutex::Autolock _l(mLock); 4778 checkForNewParameters_l(); 4779 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4780 if (!mStandby) { 4781 mInput->stream->common.standby(&mInput->stream->common); 4782 mStandby = true; 4783 } 4784 4785 if (exitPending()) break; 4786 4787 releaseWakeLock_l(); 4788 ALOGV("RecordThread: loop stopping"); 4789 // go to sleep 4790 mWaitWorkCV.wait(mLock); 4791 ALOGV("RecordThread: loop starting"); 4792 acquireWakeLock_l(); 4793 continue; 4794 } 4795 if (mActiveTrack != 0) { 4796 if (mActiveTrack->mState == TrackBase::PAUSING) { 4797 if (!mStandby) { 4798 mInput->stream->common.standby(&mInput->stream->common); 4799 mStandby = true; 4800 } 4801 mActiveTrack.clear(); 4802 mStartStopCond.broadcast(); 4803 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4804 if (mReqChannelCount != mActiveTrack->channelCount()) { 4805 mActiveTrack.clear(); 4806 mStartStopCond.broadcast(); 4807 } else if (mBytesRead != 0) { 4808 // record start succeeds only if first read from audio input 4809 // succeeds 4810 if (mBytesRead > 0) { 4811 mActiveTrack->mState = TrackBase::ACTIVE; 4812 } else { 4813 mActiveTrack.clear(); 4814 } 4815 mStartStopCond.broadcast(); 4816 } 4817 mStandby = false; 4818 } 4819 } 4820 lockEffectChains_l(effectChains); 4821 } 4822 4823 if (mActiveTrack != 0) { 4824 if (mActiveTrack->mState != TrackBase::ACTIVE && 4825 mActiveTrack->mState != TrackBase::RESUMING) { 4826 unlockEffectChains(effectChains); 4827 usleep(kRecordThreadSleepUs); 4828 continue; 4829 } 4830 for (size_t i = 0; i < effectChains.size(); i ++) { 4831 effectChains[i]->process_l(); 4832 } 4833 4834 buffer.frameCount = mFrameCount; 4835 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4836 size_t framesOut = buffer.frameCount; 4837 if (mResampler == NULL) { 4838 // no resampling 4839 while (framesOut) { 4840 size_t framesIn = mFrameCount - mRsmpInIndex; 4841 if (framesIn) { 4842 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4843 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4844 if (framesIn > framesOut) 4845 framesIn = framesOut; 4846 mRsmpInIndex += framesIn; 4847 framesOut -= framesIn; 4848 if ((int)mChannelCount == mReqChannelCount || 4849 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4850 memcpy(dst, src, framesIn * mFrameSize); 4851 } else { 4852 int16_t *src16 = (int16_t *)src; 4853 int16_t *dst16 = (int16_t *)dst; 4854 if (mChannelCount == 1) { 4855 while (framesIn--) { 4856 *dst16++ = *src16; 4857 *dst16++ = *src16++; 4858 } 4859 } else { 4860 while (framesIn--) { 4861 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4862 src16 += 2; 4863 } 4864 } 4865 } 4866 } 4867 if (framesOut && mFrameCount == mRsmpInIndex) { 4868 if (framesOut == mFrameCount && 4869 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4870 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4871 framesOut = 0; 4872 } else { 4873 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4874 mRsmpInIndex = 0; 4875 } 4876 if (mBytesRead < 0) { 4877 ALOGE("Error reading audio input"); 4878 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4879 // Force input into standby so that it tries to 4880 // recover at next read attempt 4881 mInput->stream->common.standby(&mInput->stream->common); 4882 usleep(kRecordThreadSleepUs); 4883 } 4884 mRsmpInIndex = mFrameCount; 4885 framesOut = 0; 4886 buffer.frameCount = 0; 4887 } 4888 } 4889 } 4890 } else { 4891 // resampling 4892 4893 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4894 // alter output frame count as if we were expecting stereo samples 4895 if (mChannelCount == 1 && mReqChannelCount == 1) { 4896 framesOut >>= 1; 4897 } 4898 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4899 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4900 // are 32 bit aligned which should be always true. 4901 if (mChannelCount == 2 && mReqChannelCount == 1) { 4902 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4903 // the resampler always outputs stereo samples: do post stereo to mono conversion 4904 int16_t *src = (int16_t *)mRsmpOutBuffer; 4905 int16_t *dst = buffer.i16; 4906 while (framesOut--) { 4907 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4908 src += 2; 4909 } 4910 } else { 4911 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4912 } 4913 4914 } 4915 mActiveTrack->releaseBuffer(&buffer); 4916 mActiveTrack->overflow(); 4917 } 4918 // client isn't retrieving buffers fast enough 4919 else { 4920 if (!mActiveTrack->setOverflow()) { 4921 nsecs_t now = systemTime(); 4922 if ((now - lastWarning) > kWarningThrottleNs) { 4923 ALOGW("RecordThread: buffer overflow"); 4924 lastWarning = now; 4925 } 4926 } 4927 // Release the processor for a while before asking for a new buffer. 4928 // This will give the application more chance to read from the buffer and 4929 // clear the overflow. 4930 usleep(kRecordThreadSleepUs); 4931 } 4932 } 4933 // enable changes in effect chain 4934 unlockEffectChains(effectChains); 4935 effectChains.clear(); 4936 } 4937 4938 if (!mStandby) { 4939 mInput->stream->common.standby(&mInput->stream->common); 4940 } 4941 mActiveTrack.clear(); 4942 4943 mStartStopCond.broadcast(); 4944 4945 releaseWakeLock(); 4946 4947 ALOGV("RecordThread %p exiting", this); 4948 return false; 4949} 4950 4951 4952sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4953 const sp<AudioFlinger::Client>& client, 4954 uint32_t sampleRate, 4955 audio_format_t format, 4956 int channelMask, 4957 int frameCount, 4958 int sessionId, 4959 status_t *status) 4960{ 4961 sp<RecordTrack> track; 4962 status_t lStatus; 4963 4964 lStatus = initCheck(); 4965 if (lStatus != NO_ERROR) { 4966 ALOGE("Audio driver not initialized."); 4967 goto Exit; 4968 } 4969 4970 { // scope for mLock 4971 Mutex::Autolock _l(mLock); 4972 4973 track = new RecordTrack(this, client, sampleRate, 4974 format, channelMask, frameCount, sessionId); 4975 4976 if (track->getCblk() == 0) { 4977 lStatus = NO_MEMORY; 4978 goto Exit; 4979 } 4980 4981 mTrack = track.get(); 4982 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4983 bool suspend = audio_is_bluetooth_sco_device( 4984 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4985 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4986 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4987 } 4988 lStatus = NO_ERROR; 4989 4990Exit: 4991 if (status) { 4992 *status = lStatus; 4993 } 4994 return track; 4995} 4996 4997status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4998{ 4999 ALOGV("RecordThread::start tid=%d", tid); 5000 sp <ThreadBase> strongMe = this; 5001 status_t status = NO_ERROR; 5002 { 5003 AutoMutex lock(mLock); 5004 if (mActiveTrack != 0) { 5005 if (recordTrack != mActiveTrack.get()) { 5006 status = -EBUSY; 5007 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5008 mActiveTrack->mState = TrackBase::ACTIVE; 5009 } 5010 return status; 5011 } 5012 5013 recordTrack->mState = TrackBase::IDLE; 5014 mActiveTrack = recordTrack; 5015 mLock.unlock(); 5016 status_t status = AudioSystem::startInput(mId); 5017 mLock.lock(); 5018 if (status != NO_ERROR) { 5019 mActiveTrack.clear(); 5020 return status; 5021 } 5022 mRsmpInIndex = mFrameCount; 5023 mBytesRead = 0; 5024 if (mResampler != NULL) { 5025 mResampler->reset(); 5026 } 5027 mActiveTrack->mState = TrackBase::RESUMING; 5028 // signal thread to start 5029 ALOGV("Signal record thread"); 5030 mWaitWorkCV.signal(); 5031 // do not wait for mStartStopCond if exiting 5032 if (exitPending()) { 5033 mActiveTrack.clear(); 5034 status = INVALID_OPERATION; 5035 goto startError; 5036 } 5037 mStartStopCond.wait(mLock); 5038 if (mActiveTrack == 0) { 5039 ALOGV("Record failed to start"); 5040 status = BAD_VALUE; 5041 goto startError; 5042 } 5043 ALOGV("Record started OK"); 5044 return status; 5045 } 5046startError: 5047 AudioSystem::stopInput(mId); 5048 return status; 5049} 5050 5051void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5052 ALOGV("RecordThread::stop"); 5053 sp <ThreadBase> strongMe = this; 5054 { 5055 AutoMutex lock(mLock); 5056 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5057 mActiveTrack->mState = TrackBase::PAUSING; 5058 // do not wait for mStartStopCond if exiting 5059 if (exitPending()) { 5060 return; 5061 } 5062 mStartStopCond.wait(mLock); 5063 // if we have been restarted, recordTrack == mActiveTrack.get() here 5064 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5065 mLock.unlock(); 5066 AudioSystem::stopInput(mId); 5067 mLock.lock(); 5068 ALOGV("Record stopped OK"); 5069 } 5070 } 5071 } 5072} 5073 5074status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5075{ 5076 const size_t SIZE = 256; 5077 char buffer[SIZE]; 5078 String8 result; 5079 5080 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5081 result.append(buffer); 5082 5083 if (mActiveTrack != 0) { 5084 result.append("Active Track:\n"); 5085 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5086 mActiveTrack->dump(buffer, SIZE); 5087 result.append(buffer); 5088 5089 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5090 result.append(buffer); 5091 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5092 result.append(buffer); 5093 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5094 result.append(buffer); 5095 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5096 result.append(buffer); 5097 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5098 result.append(buffer); 5099 5100 5101 } else { 5102 result.append("No record client\n"); 5103 } 5104 write(fd, result.string(), result.size()); 5105 5106 dumpBase(fd, args); 5107 dumpEffectChains(fd, args); 5108 5109 return NO_ERROR; 5110} 5111 5112// AudioBufferProvider interface 5113status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5114{ 5115 size_t framesReq = buffer->frameCount; 5116 size_t framesReady = mFrameCount - mRsmpInIndex; 5117 int channelCount; 5118 5119 if (framesReady == 0) { 5120 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5121 if (mBytesRead < 0) { 5122 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5123 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5124 // Force input into standby so that it tries to 5125 // recover at next read attempt 5126 mInput->stream->common.standby(&mInput->stream->common); 5127 usleep(kRecordThreadSleepUs); 5128 } 5129 buffer->raw = NULL; 5130 buffer->frameCount = 0; 5131 return NOT_ENOUGH_DATA; 5132 } 5133 mRsmpInIndex = 0; 5134 framesReady = mFrameCount; 5135 } 5136 5137 if (framesReq > framesReady) { 5138 framesReq = framesReady; 5139 } 5140 5141 if (mChannelCount == 1 && mReqChannelCount == 2) { 5142 channelCount = 1; 5143 } else { 5144 channelCount = 2; 5145 } 5146 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5147 buffer->frameCount = framesReq; 5148 return NO_ERROR; 5149} 5150 5151// AudioBufferProvider interface 5152void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5153{ 5154 mRsmpInIndex += buffer->frameCount; 5155 buffer->frameCount = 0; 5156} 5157 5158bool AudioFlinger::RecordThread::checkForNewParameters_l() 5159{ 5160 bool reconfig = false; 5161 5162 while (!mNewParameters.isEmpty()) { 5163 status_t status = NO_ERROR; 5164 String8 keyValuePair = mNewParameters[0]; 5165 AudioParameter param = AudioParameter(keyValuePair); 5166 int value; 5167 audio_format_t reqFormat = mFormat; 5168 int reqSamplingRate = mReqSampleRate; 5169 int reqChannelCount = mReqChannelCount; 5170 5171 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5172 reqSamplingRate = value; 5173 reconfig = true; 5174 } 5175 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5176 reqFormat = (audio_format_t) value; 5177 reconfig = true; 5178 } 5179 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5180 reqChannelCount = popcount(value); 5181 reconfig = true; 5182 } 5183 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5184 // do not accept frame count changes if tracks are open as the track buffer 5185 // size depends on frame count and correct behavior would not be guaranteed 5186 // if frame count is changed after track creation 5187 if (mActiveTrack != 0) { 5188 status = INVALID_OPERATION; 5189 } else { 5190 reconfig = true; 5191 } 5192 } 5193 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5194 // forward device change to effects that have requested to be 5195 // aware of attached audio device. 5196 for (size_t i = 0; i < mEffectChains.size(); i++) { 5197 mEffectChains[i]->setDevice_l(value); 5198 } 5199 // store input device and output device but do not forward output device to audio HAL. 5200 // Note that status is ignored by the caller for output device 5201 // (see AudioFlinger::setParameters() 5202 if (value & AUDIO_DEVICE_OUT_ALL) { 5203 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5204 status = BAD_VALUE; 5205 } else { 5206 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5207 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5208 if (mTrack != NULL) { 5209 bool suspend = audio_is_bluetooth_sco_device( 5210 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5211 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5212 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5213 } 5214 } 5215 mDevice |= (uint32_t)value; 5216 } 5217 if (status == NO_ERROR) { 5218 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5219 if (status == INVALID_OPERATION) { 5220 mInput->stream->common.standby(&mInput->stream->common); 5221 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5222 } 5223 if (reconfig) { 5224 if (status == BAD_VALUE && 5225 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5226 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5227 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5228 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5229 (reqChannelCount < 3)) { 5230 status = NO_ERROR; 5231 } 5232 if (status == NO_ERROR) { 5233 readInputParameters(); 5234 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5235 } 5236 } 5237 } 5238 5239 mNewParameters.removeAt(0); 5240 5241 mParamStatus = status; 5242 mParamCond.signal(); 5243 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5244 // already timed out waiting for the status and will never signal the condition. 5245 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5246 } 5247 return reconfig; 5248} 5249 5250String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5251{ 5252 char *s; 5253 String8 out_s8 = String8(); 5254 5255 Mutex::Autolock _l(mLock); 5256 if (initCheck() != NO_ERROR) { 5257 return out_s8; 5258 } 5259 5260 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5261 out_s8 = String8(s); 5262 free(s); 5263 return out_s8; 5264} 5265 5266void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5267 AudioSystem::OutputDescriptor desc; 5268 void *param2 = NULL; 5269 5270 switch (event) { 5271 case AudioSystem::INPUT_OPENED: 5272 case AudioSystem::INPUT_CONFIG_CHANGED: 5273 desc.channels = mChannelMask; 5274 desc.samplingRate = mSampleRate; 5275 desc.format = mFormat; 5276 desc.frameCount = mFrameCount; 5277 desc.latency = 0; 5278 param2 = &desc; 5279 break; 5280 5281 case AudioSystem::INPUT_CLOSED: 5282 default: 5283 break; 5284 } 5285 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5286} 5287 5288void AudioFlinger::RecordThread::readInputParameters() 5289{ 5290 delete mRsmpInBuffer; 5291 // mRsmpInBuffer is always assigned a new[] below 5292 delete mRsmpOutBuffer; 5293 mRsmpOutBuffer = NULL; 5294 delete mResampler; 5295 mResampler = NULL; 5296 5297 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5298 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5299 mChannelCount = (uint16_t)popcount(mChannelMask); 5300 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5301 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5302 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5303 mFrameCount = mInputBytes / mFrameSize; 5304 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5305 5306 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5307 { 5308 int channelCount; 5309 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5310 // stereo to mono post process as the resampler always outputs stereo. 5311 if (mChannelCount == 1 && mReqChannelCount == 2) { 5312 channelCount = 1; 5313 } else { 5314 channelCount = 2; 5315 } 5316 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5317 mResampler->setSampleRate(mSampleRate); 5318 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5319 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5320 5321 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5322 if (mChannelCount == 1 && mReqChannelCount == 1) { 5323 mFrameCount >>= 1; 5324 } 5325 5326 } 5327 mRsmpInIndex = mFrameCount; 5328} 5329 5330unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5331{ 5332 Mutex::Autolock _l(mLock); 5333 if (initCheck() != NO_ERROR) { 5334 return 0; 5335 } 5336 5337 return mInput->stream->get_input_frames_lost(mInput->stream); 5338} 5339 5340uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5341{ 5342 Mutex::Autolock _l(mLock); 5343 uint32_t result = 0; 5344 if (getEffectChain_l(sessionId) != 0) { 5345 result = EFFECT_SESSION; 5346 } 5347 5348 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5349 result |= TRACK_SESSION; 5350 } 5351 5352 return result; 5353} 5354 5355AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5356{ 5357 Mutex::Autolock _l(mLock); 5358 return mTrack; 5359} 5360 5361AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5362{ 5363 Mutex::Autolock _l(mLock); 5364 return mInput; 5365} 5366 5367AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5368{ 5369 Mutex::Autolock _l(mLock); 5370 AudioStreamIn *input = mInput; 5371 mInput = NULL; 5372 return input; 5373} 5374 5375// this method must always be called either with ThreadBase mLock held or inside the thread loop 5376audio_stream_t* AudioFlinger::RecordThread::stream() 5377{ 5378 if (mInput == NULL) { 5379 return NULL; 5380 } 5381 return &mInput->stream->common; 5382} 5383 5384 5385// ---------------------------------------------------------------------------- 5386 5387audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5388 uint32_t *pSamplingRate, 5389 audio_format_t *pFormat, 5390 uint32_t *pChannels, 5391 uint32_t *pLatencyMs, 5392 uint32_t flags) 5393{ 5394 status_t status; 5395 PlaybackThread *thread = NULL; 5396 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5397 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5398 uint32_t channels = pChannels ? *pChannels : 0; 5399 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5400 audio_stream_out_t *outStream; 5401 audio_hw_device_t *outHwDev; 5402 5403 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5404 pDevices ? *pDevices : 0, 5405 samplingRate, 5406 format, 5407 channels, 5408 flags); 5409 5410 if (pDevices == NULL || *pDevices == 0) { 5411 return 0; 5412 } 5413 5414 Mutex::Autolock _l(mLock); 5415 5416 outHwDev = findSuitableHwDev_l(*pDevices); 5417 if (outHwDev == NULL) 5418 return 0; 5419 5420 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5421 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5422 &channels, &samplingRate, &outStream); 5423 mHardwareStatus = AUDIO_HW_IDLE; 5424 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5425 outStream, 5426 samplingRate, 5427 format, 5428 channels, 5429 status); 5430 5431 if (outStream != NULL) { 5432 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5433 audio_io_handle_t id = nextUniqueId(); 5434 5435 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5436 (format != AUDIO_FORMAT_PCM_16_BIT) || 5437 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5438 thread = new DirectOutputThread(this, output, id, *pDevices); 5439 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5440 } else { 5441 thread = new MixerThread(this, output, id, *pDevices); 5442 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5443 } 5444 mPlaybackThreads.add(id, thread); 5445 5446 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5447 if (pFormat != NULL) *pFormat = format; 5448 if (pChannels != NULL) *pChannels = channels; 5449 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5450 5451 // notify client processes of the new output creation 5452 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5453 return id; 5454 } 5455 5456 return 0; 5457} 5458 5459audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5460 audio_io_handle_t output2) 5461{ 5462 Mutex::Autolock _l(mLock); 5463 MixerThread *thread1 = checkMixerThread_l(output1); 5464 MixerThread *thread2 = checkMixerThread_l(output2); 5465 5466 if (thread1 == NULL || thread2 == NULL) { 5467 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5468 return 0; 5469 } 5470 5471 audio_io_handle_t id = nextUniqueId(); 5472 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5473 thread->addOutputTrack(thread2); 5474 mPlaybackThreads.add(id, thread); 5475 // notify client processes of the new output creation 5476 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5477 return id; 5478} 5479 5480status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5481{ 5482 // keep strong reference on the playback thread so that 5483 // it is not destroyed while exit() is executed 5484 sp <PlaybackThread> thread; 5485 { 5486 Mutex::Autolock _l(mLock); 5487 thread = checkPlaybackThread_l(output); 5488 if (thread == NULL) { 5489 return BAD_VALUE; 5490 } 5491 5492 ALOGV("closeOutput() %d", output); 5493 5494 if (thread->type() == ThreadBase::MIXER) { 5495 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5496 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5497 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5498 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5499 } 5500 } 5501 } 5502 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5503 mPlaybackThreads.removeItem(output); 5504 } 5505 thread->exit(); 5506 // The thread entity (active unit of execution) is no longer running here, 5507 // but the ThreadBase container still exists. 5508 5509 if (thread->type() != ThreadBase::DUPLICATING) { 5510 AudioStreamOut *out = thread->clearOutput(); 5511 assert(out != NULL); 5512 // from now on thread->mOutput is NULL 5513 out->hwDev->close_output_stream(out->hwDev, out->stream); 5514 delete out; 5515 } 5516 return NO_ERROR; 5517} 5518 5519status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5520{ 5521 Mutex::Autolock _l(mLock); 5522 PlaybackThread *thread = checkPlaybackThread_l(output); 5523 5524 if (thread == NULL) { 5525 return BAD_VALUE; 5526 } 5527 5528 ALOGV("suspendOutput() %d", output); 5529 thread->suspend(); 5530 5531 return NO_ERROR; 5532} 5533 5534status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5535{ 5536 Mutex::Autolock _l(mLock); 5537 PlaybackThread *thread = checkPlaybackThread_l(output); 5538 5539 if (thread == NULL) { 5540 return BAD_VALUE; 5541 } 5542 5543 ALOGV("restoreOutput() %d", output); 5544 5545 thread->restore(); 5546 5547 return NO_ERROR; 5548} 5549 5550audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5551 uint32_t *pSamplingRate, 5552 audio_format_t *pFormat, 5553 uint32_t *pChannels, 5554 audio_in_acoustics_t acoustics) 5555{ 5556 status_t status; 5557 RecordThread *thread = NULL; 5558 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5559 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5560 uint32_t channels = pChannels ? *pChannels : 0; 5561 uint32_t reqSamplingRate = samplingRate; 5562 audio_format_t reqFormat = format; 5563 uint32_t reqChannels = channels; 5564 audio_stream_in_t *inStream; 5565 audio_hw_device_t *inHwDev; 5566 5567 if (pDevices == NULL || *pDevices == 0) { 5568 return 0; 5569 } 5570 5571 Mutex::Autolock _l(mLock); 5572 5573 inHwDev = findSuitableHwDev_l(*pDevices); 5574 if (inHwDev == NULL) 5575 return 0; 5576 5577 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5578 &channels, &samplingRate, 5579 acoustics, 5580 &inStream); 5581 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5582 inStream, 5583 samplingRate, 5584 format, 5585 channels, 5586 acoustics, 5587 status); 5588 5589 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5590 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5591 // or stereo to mono conversions on 16 bit PCM inputs. 5592 if (inStream == NULL && status == BAD_VALUE && 5593 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5594 (samplingRate <= 2 * reqSamplingRate) && 5595 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5596 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5597 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5598 &channels, &samplingRate, 5599 acoustics, 5600 &inStream); 5601 } 5602 5603 if (inStream != NULL) { 5604 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5605 5606 audio_io_handle_t id = nextUniqueId(); 5607 // Start record thread 5608 // RecorThread require both input and output device indication to forward to audio 5609 // pre processing modules 5610 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5611 thread = new RecordThread(this, 5612 input, 5613 reqSamplingRate, 5614 reqChannels, 5615 id, 5616 device); 5617 mRecordThreads.add(id, thread); 5618 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5619 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5620 if (pFormat != NULL) *pFormat = format; 5621 if (pChannels != NULL) *pChannels = reqChannels; 5622 5623 input->stream->common.standby(&input->stream->common); 5624 5625 // notify client processes of the new input creation 5626 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5627 return id; 5628 } 5629 5630 return 0; 5631} 5632 5633status_t AudioFlinger::closeInput(audio_io_handle_t input) 5634{ 5635 // keep strong reference on the record thread so that 5636 // it is not destroyed while exit() is executed 5637 sp <RecordThread> thread; 5638 { 5639 Mutex::Autolock _l(mLock); 5640 thread = checkRecordThread_l(input); 5641 if (thread == NULL) { 5642 return BAD_VALUE; 5643 } 5644 5645 ALOGV("closeInput() %d", input); 5646 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5647 mRecordThreads.removeItem(input); 5648 } 5649 thread->exit(); 5650 // The thread entity (active unit of execution) is no longer running here, 5651 // but the ThreadBase container still exists. 5652 5653 AudioStreamIn *in = thread->clearInput(); 5654 assert(in != NULL); 5655 // from now on thread->mInput is NULL 5656 in->hwDev->close_input_stream(in->hwDev, in->stream); 5657 delete in; 5658 5659 return NO_ERROR; 5660} 5661 5662status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5663{ 5664 Mutex::Autolock _l(mLock); 5665 MixerThread *dstThread = checkMixerThread_l(output); 5666 if (dstThread == NULL) { 5667 ALOGW("setStreamOutput() bad output id %d", output); 5668 return BAD_VALUE; 5669 } 5670 5671 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5672 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5673 5674 dstThread->setStreamValid(stream, true); 5675 5676 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5677 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5678 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5679 MixerThread *srcThread = (MixerThread *)thread; 5680 srcThread->setStreamValid(stream, false); 5681 srcThread->invalidateTracks(stream); 5682 } 5683 } 5684 5685 return NO_ERROR; 5686} 5687 5688 5689int AudioFlinger::newAudioSessionId() 5690{ 5691 return nextUniqueId(); 5692} 5693 5694void AudioFlinger::acquireAudioSessionId(int audioSession) 5695{ 5696 Mutex::Autolock _l(mLock); 5697 pid_t caller = IPCThreadState::self()->getCallingPid(); 5698 ALOGV("acquiring %d from %d", audioSession, caller); 5699 size_t num = mAudioSessionRefs.size(); 5700 for (size_t i = 0; i< num; i++) { 5701 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5702 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5703 ref->mCnt++; 5704 ALOGV(" incremented refcount to %d", ref->mCnt); 5705 return; 5706 } 5707 } 5708 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5709 ALOGV(" added new entry for %d", audioSession); 5710} 5711 5712void AudioFlinger::releaseAudioSessionId(int audioSession) 5713{ 5714 Mutex::Autolock _l(mLock); 5715 pid_t caller = IPCThreadState::self()->getCallingPid(); 5716 ALOGV("releasing %d from %d", audioSession, caller); 5717 size_t num = mAudioSessionRefs.size(); 5718 for (size_t i = 0; i< num; i++) { 5719 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5720 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5721 ref->mCnt--; 5722 ALOGV(" decremented refcount to %d", ref->mCnt); 5723 if (ref->mCnt == 0) { 5724 mAudioSessionRefs.removeAt(i); 5725 delete ref; 5726 purgeStaleEffects_l(); 5727 } 5728 return; 5729 } 5730 } 5731 ALOGW("session id %d not found for pid %d", audioSession, caller); 5732} 5733 5734void AudioFlinger::purgeStaleEffects_l() { 5735 5736 ALOGV("purging stale effects"); 5737 5738 Vector< sp<EffectChain> > chains; 5739 5740 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5741 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5742 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5743 sp<EffectChain> ec = t->mEffectChains[j]; 5744 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5745 chains.push(ec); 5746 } 5747 } 5748 } 5749 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5750 sp<RecordThread> t = mRecordThreads.valueAt(i); 5751 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5752 sp<EffectChain> ec = t->mEffectChains[j]; 5753 chains.push(ec); 5754 } 5755 } 5756 5757 for (size_t i = 0; i < chains.size(); i++) { 5758 sp<EffectChain> ec = chains[i]; 5759 int sessionid = ec->sessionId(); 5760 sp<ThreadBase> t = ec->mThread.promote(); 5761 if (t == 0) { 5762 continue; 5763 } 5764 size_t numsessionrefs = mAudioSessionRefs.size(); 5765 bool found = false; 5766 for (size_t k = 0; k < numsessionrefs; k++) { 5767 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5768 if (ref->mSessionid == sessionid) { 5769 ALOGV(" session %d still exists for %d with %d refs", 5770 sessionid, ref->mPid, ref->mCnt); 5771 found = true; 5772 break; 5773 } 5774 } 5775 if (!found) { 5776 // remove all effects from the chain 5777 while (ec->mEffects.size()) { 5778 sp<EffectModule> effect = ec->mEffects[0]; 5779 effect->unPin(); 5780 Mutex::Autolock _l (t->mLock); 5781 t->removeEffect_l(effect); 5782 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5783 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5784 if (handle != 0) { 5785 handle->mEffect.clear(); 5786 if (handle->mHasControl && handle->mEnabled) { 5787 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5788 } 5789 } 5790 } 5791 AudioSystem::unregisterEffect(effect->id()); 5792 } 5793 } 5794 } 5795 return; 5796} 5797 5798// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5799AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5800{ 5801 return mPlaybackThreads.valueFor(output).get(); 5802} 5803 5804// checkMixerThread_l() must be called with AudioFlinger::mLock held 5805AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5806{ 5807 PlaybackThread *thread = checkPlaybackThread_l(output); 5808 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5809} 5810 5811// checkRecordThread_l() must be called with AudioFlinger::mLock held 5812AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5813{ 5814 return mRecordThreads.valueFor(input).get(); 5815} 5816 5817uint32_t AudioFlinger::nextUniqueId() 5818{ 5819 return android_atomic_inc(&mNextUniqueId); 5820} 5821 5822AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5823{ 5824 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5825 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5826 AudioStreamOut *output = thread->getOutput(); 5827 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5828 return thread; 5829 } 5830 } 5831 return NULL; 5832} 5833 5834uint32_t AudioFlinger::primaryOutputDevice_l() const 5835{ 5836 PlaybackThread *thread = primaryPlaybackThread_l(); 5837 5838 if (thread == NULL) { 5839 return 0; 5840 } 5841 5842 return thread->device(); 5843} 5844 5845 5846// ---------------------------------------------------------------------------- 5847// Effect management 5848// ---------------------------------------------------------------------------- 5849 5850 5851status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5852{ 5853 Mutex::Autolock _l(mLock); 5854 return EffectQueryNumberEffects(numEffects); 5855} 5856 5857status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5858{ 5859 Mutex::Autolock _l(mLock); 5860 return EffectQueryEffect(index, descriptor); 5861} 5862 5863status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5864 effect_descriptor_t *descriptor) const 5865{ 5866 Mutex::Autolock _l(mLock); 5867 return EffectGetDescriptor(pUuid, descriptor); 5868} 5869 5870 5871sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5872 effect_descriptor_t *pDesc, 5873 const sp<IEffectClient>& effectClient, 5874 int32_t priority, 5875 audio_io_handle_t io, 5876 int sessionId, 5877 status_t *status, 5878 int *id, 5879 int *enabled) 5880{ 5881 status_t lStatus = NO_ERROR; 5882 sp<EffectHandle> handle; 5883 effect_descriptor_t desc; 5884 5885 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5886 pid, effectClient.get(), priority, sessionId, io); 5887 5888 if (pDesc == NULL) { 5889 lStatus = BAD_VALUE; 5890 goto Exit; 5891 } 5892 5893 // check audio settings permission for global effects 5894 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5895 lStatus = PERMISSION_DENIED; 5896 goto Exit; 5897 } 5898 5899 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5900 // that can only be created by audio policy manager (running in same process) 5901 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5902 lStatus = PERMISSION_DENIED; 5903 goto Exit; 5904 } 5905 5906 if (io == 0) { 5907 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5908 // output must be specified by AudioPolicyManager when using session 5909 // AUDIO_SESSION_OUTPUT_STAGE 5910 lStatus = BAD_VALUE; 5911 goto Exit; 5912 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5913 // if the output returned by getOutputForEffect() is removed before we lock the 5914 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5915 // and we will exit safely 5916 io = AudioSystem::getOutputForEffect(&desc); 5917 } 5918 } 5919 5920 { 5921 Mutex::Autolock _l(mLock); 5922 5923 5924 if (!EffectIsNullUuid(&pDesc->uuid)) { 5925 // if uuid is specified, request effect descriptor 5926 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5927 if (lStatus < 0) { 5928 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5929 goto Exit; 5930 } 5931 } else { 5932 // if uuid is not specified, look for an available implementation 5933 // of the required type in effect factory 5934 if (EffectIsNullUuid(&pDesc->type)) { 5935 ALOGW("createEffect() no effect type"); 5936 lStatus = BAD_VALUE; 5937 goto Exit; 5938 } 5939 uint32_t numEffects = 0; 5940 effect_descriptor_t d; 5941 d.flags = 0; // prevent compiler warning 5942 bool found = false; 5943 5944 lStatus = EffectQueryNumberEffects(&numEffects); 5945 if (lStatus < 0) { 5946 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5947 goto Exit; 5948 } 5949 for (uint32_t i = 0; i < numEffects; i++) { 5950 lStatus = EffectQueryEffect(i, &desc); 5951 if (lStatus < 0) { 5952 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5953 continue; 5954 } 5955 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5956 // If matching type found save effect descriptor. If the session is 5957 // 0 and the effect is not auxiliary, continue enumeration in case 5958 // an auxiliary version of this effect type is available 5959 found = true; 5960 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5961 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5962 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5963 break; 5964 } 5965 } 5966 } 5967 if (!found) { 5968 lStatus = BAD_VALUE; 5969 ALOGW("createEffect() effect not found"); 5970 goto Exit; 5971 } 5972 // For same effect type, chose auxiliary version over insert version if 5973 // connect to output mix (Compliance to OpenSL ES) 5974 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5975 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5976 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5977 } 5978 } 5979 5980 // Do not allow auxiliary effects on a session different from 0 (output mix) 5981 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5982 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5983 lStatus = INVALID_OPERATION; 5984 goto Exit; 5985 } 5986 5987 // check recording permission for visualizer 5988 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5989 !recordingAllowed()) { 5990 lStatus = PERMISSION_DENIED; 5991 goto Exit; 5992 } 5993 5994 // return effect descriptor 5995 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5996 5997 // If output is not specified try to find a matching audio session ID in one of the 5998 // output threads. 5999 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6000 // because of code checking output when entering the function. 6001 // Note: io is never 0 when creating an effect on an input 6002 if (io == 0) { 6003 // look for the thread where the specified audio session is present 6004 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6005 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6006 io = mPlaybackThreads.keyAt(i); 6007 break; 6008 } 6009 } 6010 if (io == 0) { 6011 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6012 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6013 io = mRecordThreads.keyAt(i); 6014 break; 6015 } 6016 } 6017 } 6018 // If no output thread contains the requested session ID, default to 6019 // first output. The effect chain will be moved to the correct output 6020 // thread when a track with the same session ID is created 6021 if (io == 0 && mPlaybackThreads.size()) { 6022 io = mPlaybackThreads.keyAt(0); 6023 } 6024 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6025 } 6026 ThreadBase *thread = checkRecordThread_l(io); 6027 if (thread == NULL) { 6028 thread = checkPlaybackThread_l(io); 6029 if (thread == NULL) { 6030 ALOGE("createEffect() unknown output thread"); 6031 lStatus = BAD_VALUE; 6032 goto Exit; 6033 } 6034 } 6035 6036 sp<Client> client = registerPid_l(pid); 6037 6038 // create effect on selected output thread 6039 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6040 &desc, enabled, &lStatus); 6041 if (handle != 0 && id != NULL) { 6042 *id = handle->id(); 6043 } 6044 } 6045 6046Exit: 6047 if(status) { 6048 *status = lStatus; 6049 } 6050 return handle; 6051} 6052 6053status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6054 audio_io_handle_t dstOutput) 6055{ 6056 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6057 sessionId, srcOutput, dstOutput); 6058 Mutex::Autolock _l(mLock); 6059 if (srcOutput == dstOutput) { 6060 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6061 return NO_ERROR; 6062 } 6063 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6064 if (srcThread == NULL) { 6065 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6066 return BAD_VALUE; 6067 } 6068 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6069 if (dstThread == NULL) { 6070 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6071 return BAD_VALUE; 6072 } 6073 6074 Mutex::Autolock _dl(dstThread->mLock); 6075 Mutex::Autolock _sl(srcThread->mLock); 6076 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6077 6078 return NO_ERROR; 6079} 6080 6081// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6082status_t AudioFlinger::moveEffectChain_l(int sessionId, 6083 AudioFlinger::PlaybackThread *srcThread, 6084 AudioFlinger::PlaybackThread *dstThread, 6085 bool reRegister) 6086{ 6087 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6088 sessionId, srcThread, dstThread); 6089 6090 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6091 if (chain == 0) { 6092 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6093 sessionId, srcThread); 6094 return INVALID_OPERATION; 6095 } 6096 6097 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6098 // so that a new chain is created with correct parameters when first effect is added. This is 6099 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6100 // removed. 6101 srcThread->removeEffectChain_l(chain); 6102 6103 // transfer all effects one by one so that new effect chain is created on new thread with 6104 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6105 audio_io_handle_t dstOutput = dstThread->id(); 6106 sp<EffectChain> dstChain; 6107 uint32_t strategy = 0; // prevent compiler warning 6108 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6109 while (effect != 0) { 6110 srcThread->removeEffect_l(effect); 6111 dstThread->addEffect_l(effect); 6112 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6113 if (effect->state() == EffectModule::ACTIVE || 6114 effect->state() == EffectModule::STOPPING) { 6115 effect->start(); 6116 } 6117 // if the move request is not received from audio policy manager, the effect must be 6118 // re-registered with the new strategy and output 6119 if (dstChain == 0) { 6120 dstChain = effect->chain().promote(); 6121 if (dstChain == 0) { 6122 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6123 srcThread->addEffect_l(effect); 6124 return NO_INIT; 6125 } 6126 strategy = dstChain->strategy(); 6127 } 6128 if (reRegister) { 6129 AudioSystem::unregisterEffect(effect->id()); 6130 AudioSystem::registerEffect(&effect->desc(), 6131 dstOutput, 6132 strategy, 6133 sessionId, 6134 effect->id()); 6135 } 6136 effect = chain->getEffectFromId_l(0); 6137 } 6138 6139 return NO_ERROR; 6140} 6141 6142 6143// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6144sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6145 const sp<AudioFlinger::Client>& client, 6146 const sp<IEffectClient>& effectClient, 6147 int32_t priority, 6148 int sessionId, 6149 effect_descriptor_t *desc, 6150 int *enabled, 6151 status_t *status 6152 ) 6153{ 6154 sp<EffectModule> effect; 6155 sp<EffectHandle> handle; 6156 status_t lStatus; 6157 sp<EffectChain> chain; 6158 bool chainCreated = false; 6159 bool effectCreated = false; 6160 bool effectRegistered = false; 6161 6162 lStatus = initCheck(); 6163 if (lStatus != NO_ERROR) { 6164 ALOGW("createEffect_l() Audio driver not initialized."); 6165 goto Exit; 6166 } 6167 6168 // Do not allow effects with session ID 0 on direct output or duplicating threads 6169 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6170 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6171 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6172 desc->name, sessionId); 6173 lStatus = BAD_VALUE; 6174 goto Exit; 6175 } 6176 // Only Pre processor effects are allowed on input threads and only on input threads 6177 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6178 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6179 desc->name, desc->flags, mType); 6180 lStatus = BAD_VALUE; 6181 goto Exit; 6182 } 6183 6184 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6185 6186 { // scope for mLock 6187 Mutex::Autolock _l(mLock); 6188 6189 // check for existing effect chain with the requested audio session 6190 chain = getEffectChain_l(sessionId); 6191 if (chain == 0) { 6192 // create a new chain for this session 6193 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6194 chain = new EffectChain(this, sessionId); 6195 addEffectChain_l(chain); 6196 chain->setStrategy(getStrategyForSession_l(sessionId)); 6197 chainCreated = true; 6198 } else { 6199 effect = chain->getEffectFromDesc_l(desc); 6200 } 6201 6202 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6203 6204 if (effect == 0) { 6205 int id = mAudioFlinger->nextUniqueId(); 6206 // Check CPU and memory usage 6207 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6208 if (lStatus != NO_ERROR) { 6209 goto Exit; 6210 } 6211 effectRegistered = true; 6212 // create a new effect module if none present in the chain 6213 effect = new EffectModule(this, chain, desc, id, sessionId); 6214 lStatus = effect->status(); 6215 if (lStatus != NO_ERROR) { 6216 goto Exit; 6217 } 6218 lStatus = chain->addEffect_l(effect); 6219 if (lStatus != NO_ERROR) { 6220 goto Exit; 6221 } 6222 effectCreated = true; 6223 6224 effect->setDevice(mDevice); 6225 effect->setMode(mAudioFlinger->getMode()); 6226 } 6227 // create effect handle and connect it to effect module 6228 handle = new EffectHandle(effect, client, effectClient, priority); 6229 lStatus = effect->addHandle(handle); 6230 if (enabled != NULL) { 6231 *enabled = (int)effect->isEnabled(); 6232 } 6233 } 6234 6235Exit: 6236 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6237 Mutex::Autolock _l(mLock); 6238 if (effectCreated) { 6239 chain->removeEffect_l(effect); 6240 } 6241 if (effectRegistered) { 6242 AudioSystem::unregisterEffect(effect->id()); 6243 } 6244 if (chainCreated) { 6245 removeEffectChain_l(chain); 6246 } 6247 handle.clear(); 6248 } 6249 6250 if(status) { 6251 *status = lStatus; 6252 } 6253 return handle; 6254} 6255 6256sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6257{ 6258 sp<EffectChain> chain = getEffectChain_l(sessionId); 6259 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6260} 6261 6262// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6263// PlaybackThread::mLock held 6264status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6265{ 6266 // check for existing effect chain with the requested audio session 6267 int sessionId = effect->sessionId(); 6268 sp<EffectChain> chain = getEffectChain_l(sessionId); 6269 bool chainCreated = false; 6270 6271 if (chain == 0) { 6272 // create a new chain for this session 6273 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6274 chain = new EffectChain(this, sessionId); 6275 addEffectChain_l(chain); 6276 chain->setStrategy(getStrategyForSession_l(sessionId)); 6277 chainCreated = true; 6278 } 6279 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6280 6281 if (chain->getEffectFromId_l(effect->id()) != 0) { 6282 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6283 this, effect->desc().name, chain.get()); 6284 return BAD_VALUE; 6285 } 6286 6287 status_t status = chain->addEffect_l(effect); 6288 if (status != NO_ERROR) { 6289 if (chainCreated) { 6290 removeEffectChain_l(chain); 6291 } 6292 return status; 6293 } 6294 6295 effect->setDevice(mDevice); 6296 effect->setMode(mAudioFlinger->getMode()); 6297 return NO_ERROR; 6298} 6299 6300void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6301 6302 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6303 effect_descriptor_t desc = effect->desc(); 6304 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6305 detachAuxEffect_l(effect->id()); 6306 } 6307 6308 sp<EffectChain> chain = effect->chain().promote(); 6309 if (chain != 0) { 6310 // remove effect chain if removing last effect 6311 if (chain->removeEffect_l(effect) == 0) { 6312 removeEffectChain_l(chain); 6313 } 6314 } else { 6315 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6316 } 6317} 6318 6319void AudioFlinger::ThreadBase::lockEffectChains_l( 6320 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6321{ 6322 effectChains = mEffectChains; 6323 for (size_t i = 0; i < mEffectChains.size(); i++) { 6324 mEffectChains[i]->lock(); 6325 } 6326} 6327 6328void AudioFlinger::ThreadBase::unlockEffectChains( 6329 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6330{ 6331 for (size_t i = 0; i < effectChains.size(); i++) { 6332 effectChains[i]->unlock(); 6333 } 6334} 6335 6336sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6337{ 6338 Mutex::Autolock _l(mLock); 6339 return getEffectChain_l(sessionId); 6340} 6341 6342sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6343{ 6344 size_t size = mEffectChains.size(); 6345 for (size_t i = 0; i < size; i++) { 6346 if (mEffectChains[i]->sessionId() == sessionId) { 6347 return mEffectChains[i]; 6348 } 6349 } 6350 return 0; 6351} 6352 6353void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6354{ 6355 Mutex::Autolock _l(mLock); 6356 size_t size = mEffectChains.size(); 6357 for (size_t i = 0; i < size; i++) { 6358 mEffectChains[i]->setMode_l(mode); 6359 } 6360} 6361 6362void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6363 const wp<EffectHandle>& handle, 6364 bool unpinIfLast) { 6365 6366 Mutex::Autolock _l(mLock); 6367 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6368 // delete the effect module if removing last handle on it 6369 if (effect->removeHandle(handle) == 0) { 6370 if (!effect->isPinned() || unpinIfLast) { 6371 removeEffect_l(effect); 6372 AudioSystem::unregisterEffect(effect->id()); 6373 } 6374 } 6375} 6376 6377status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6378{ 6379 int session = chain->sessionId(); 6380 int16_t *buffer = mMixBuffer; 6381 bool ownsBuffer = false; 6382 6383 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6384 if (session > 0) { 6385 // Only one effect chain can be present in direct output thread and it uses 6386 // the mix buffer as input 6387 if (mType != DIRECT) { 6388 size_t numSamples = mFrameCount * mChannelCount; 6389 buffer = new int16_t[numSamples]; 6390 memset(buffer, 0, numSamples * sizeof(int16_t)); 6391 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6392 ownsBuffer = true; 6393 } 6394 6395 // Attach all tracks with same session ID to this chain. 6396 for (size_t i = 0; i < mTracks.size(); ++i) { 6397 sp<Track> track = mTracks[i]; 6398 if (session == track->sessionId()) { 6399 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6400 track->setMainBuffer(buffer); 6401 chain->incTrackCnt(); 6402 } 6403 } 6404 6405 // indicate all active tracks in the chain 6406 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6407 sp<Track> track = mActiveTracks[i].promote(); 6408 if (track == 0) continue; 6409 if (session == track->sessionId()) { 6410 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6411 chain->incActiveTrackCnt(); 6412 } 6413 } 6414 } 6415 6416 chain->setInBuffer(buffer, ownsBuffer); 6417 chain->setOutBuffer(mMixBuffer); 6418 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6419 // chains list in order to be processed last as it contains output stage effects 6420 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6421 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6422 // after track specific effects and before output stage 6423 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6424 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6425 // Effect chain for other sessions are inserted at beginning of effect 6426 // chains list to be processed before output mix effects. Relative order between other 6427 // sessions is not important 6428 size_t size = mEffectChains.size(); 6429 size_t i = 0; 6430 for (i = 0; i < size; i++) { 6431 if (mEffectChains[i]->sessionId() < session) break; 6432 } 6433 mEffectChains.insertAt(chain, i); 6434 checkSuspendOnAddEffectChain_l(chain); 6435 6436 return NO_ERROR; 6437} 6438 6439size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6440{ 6441 int session = chain->sessionId(); 6442 6443 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6444 6445 for (size_t i = 0; i < mEffectChains.size(); i++) { 6446 if (chain == mEffectChains[i]) { 6447 mEffectChains.removeAt(i); 6448 // detach all active tracks from the chain 6449 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6450 sp<Track> track = mActiveTracks[i].promote(); 6451 if (track == 0) continue; 6452 if (session == track->sessionId()) { 6453 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6454 chain.get(), session); 6455 chain->decActiveTrackCnt(); 6456 } 6457 } 6458 6459 // detach all tracks with same session ID from this chain 6460 for (size_t i = 0; i < mTracks.size(); ++i) { 6461 sp<Track> track = mTracks[i]; 6462 if (session == track->sessionId()) { 6463 track->setMainBuffer(mMixBuffer); 6464 chain->decTrackCnt(); 6465 } 6466 } 6467 break; 6468 } 6469 } 6470 return mEffectChains.size(); 6471} 6472 6473status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6474 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6475{ 6476 Mutex::Autolock _l(mLock); 6477 return attachAuxEffect_l(track, EffectId); 6478} 6479 6480status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6481 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6482{ 6483 status_t status = NO_ERROR; 6484 6485 if (EffectId == 0) { 6486 track->setAuxBuffer(0, NULL); 6487 } else { 6488 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6489 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6490 if (effect != 0) { 6491 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6492 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6493 } else { 6494 status = INVALID_OPERATION; 6495 } 6496 } else { 6497 status = BAD_VALUE; 6498 } 6499 } 6500 return status; 6501} 6502 6503void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6504{ 6505 for (size_t i = 0; i < mTracks.size(); ++i) { 6506 sp<Track> track = mTracks[i]; 6507 if (track->auxEffectId() == effectId) { 6508 attachAuxEffect_l(track, 0); 6509 } 6510 } 6511} 6512 6513status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6514{ 6515 // only one chain per input thread 6516 if (mEffectChains.size() != 0) { 6517 return INVALID_OPERATION; 6518 } 6519 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6520 6521 chain->setInBuffer(NULL); 6522 chain->setOutBuffer(NULL); 6523 6524 checkSuspendOnAddEffectChain_l(chain); 6525 6526 mEffectChains.add(chain); 6527 6528 return NO_ERROR; 6529} 6530 6531size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6532{ 6533 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6534 ALOGW_IF(mEffectChains.size() != 1, 6535 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6536 chain.get(), mEffectChains.size(), this); 6537 if (mEffectChains.size() == 1) { 6538 mEffectChains.removeAt(0); 6539 } 6540 return 0; 6541} 6542 6543// ---------------------------------------------------------------------------- 6544// EffectModule implementation 6545// ---------------------------------------------------------------------------- 6546 6547#undef LOG_TAG 6548#define LOG_TAG "AudioFlinger::EffectModule" 6549 6550AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6551 const wp<AudioFlinger::EffectChain>& chain, 6552 effect_descriptor_t *desc, 6553 int id, 6554 int sessionId) 6555 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6556 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6557{ 6558 ALOGV("Constructor %p", this); 6559 int lStatus; 6560 if (thread == NULL) { 6561 return; 6562 } 6563 6564 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6565 6566 // create effect engine from effect factory 6567 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6568 6569 if (mStatus != NO_ERROR) { 6570 return; 6571 } 6572 lStatus = init(); 6573 if (lStatus < 0) { 6574 mStatus = lStatus; 6575 goto Error; 6576 } 6577 6578 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6579 mPinned = true; 6580 } 6581 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6582 return; 6583Error: 6584 EffectRelease(mEffectInterface); 6585 mEffectInterface = NULL; 6586 ALOGV("Constructor Error %d", mStatus); 6587} 6588 6589AudioFlinger::EffectModule::~EffectModule() 6590{ 6591 ALOGV("Destructor %p", this); 6592 if (mEffectInterface != NULL) { 6593 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6594 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6595 sp<ThreadBase> thread = mThread.promote(); 6596 if (thread != 0) { 6597 audio_stream_t *stream = thread->stream(); 6598 if (stream != NULL) { 6599 stream->remove_audio_effect(stream, mEffectInterface); 6600 } 6601 } 6602 } 6603 // release effect engine 6604 EffectRelease(mEffectInterface); 6605 } 6606} 6607 6608status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6609{ 6610 status_t status; 6611 6612 Mutex::Autolock _l(mLock); 6613 int priority = handle->priority(); 6614 size_t size = mHandles.size(); 6615 sp<EffectHandle> h; 6616 size_t i; 6617 for (i = 0; i < size; i++) { 6618 h = mHandles[i].promote(); 6619 if (h == 0) continue; 6620 if (h->priority() <= priority) break; 6621 } 6622 // if inserted in first place, move effect control from previous owner to this handle 6623 if (i == 0) { 6624 bool enabled = false; 6625 if (h != 0) { 6626 enabled = h->enabled(); 6627 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6628 } 6629 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6630 status = NO_ERROR; 6631 } else { 6632 status = ALREADY_EXISTS; 6633 } 6634 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6635 mHandles.insertAt(handle, i); 6636 return status; 6637} 6638 6639size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6640{ 6641 Mutex::Autolock _l(mLock); 6642 size_t size = mHandles.size(); 6643 size_t i; 6644 for (i = 0; i < size; i++) { 6645 if (mHandles[i] == handle) break; 6646 } 6647 if (i == size) { 6648 return size; 6649 } 6650 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6651 6652 bool enabled = false; 6653 EffectHandle *hdl = handle.unsafe_get(); 6654 if (hdl != NULL) { 6655 ALOGV("removeHandle() unsafe_get OK"); 6656 enabled = hdl->enabled(); 6657 } 6658 mHandles.removeAt(i); 6659 size = mHandles.size(); 6660 // if removed from first place, move effect control from this handle to next in line 6661 if (i == 0 && size != 0) { 6662 sp<EffectHandle> h = mHandles[0].promote(); 6663 if (h != 0) { 6664 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6665 } 6666 } 6667 6668 // Prevent calls to process() and other functions on effect interface from now on. 6669 // The effect engine will be released by the destructor when the last strong reference on 6670 // this object is released which can happen after next process is called. 6671 if (size == 0 && !mPinned) { 6672 mState = DESTROYED; 6673 } 6674 6675 return size; 6676} 6677 6678sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6679{ 6680 Mutex::Autolock _l(mLock); 6681 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6682} 6683 6684void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6685{ 6686 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6687 // keep a strong reference on this EffectModule to avoid calling the 6688 // destructor before we exit 6689 sp<EffectModule> keep(this); 6690 { 6691 sp<ThreadBase> thread = mThread.promote(); 6692 if (thread != 0) { 6693 thread->disconnectEffect(keep, handle, unpinIfLast); 6694 } 6695 } 6696} 6697 6698void AudioFlinger::EffectModule::updateState() { 6699 Mutex::Autolock _l(mLock); 6700 6701 switch (mState) { 6702 case RESTART: 6703 reset_l(); 6704 // FALL THROUGH 6705 6706 case STARTING: 6707 // clear auxiliary effect input buffer for next accumulation 6708 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6709 memset(mConfig.inputCfg.buffer.raw, 6710 0, 6711 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6712 } 6713 start_l(); 6714 mState = ACTIVE; 6715 break; 6716 case STOPPING: 6717 stop_l(); 6718 mDisableWaitCnt = mMaxDisableWaitCnt; 6719 mState = STOPPED; 6720 break; 6721 case STOPPED: 6722 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6723 // turn off sequence. 6724 if (--mDisableWaitCnt == 0) { 6725 reset_l(); 6726 mState = IDLE; 6727 } 6728 break; 6729 default: //IDLE , ACTIVE, DESTROYED 6730 break; 6731 } 6732} 6733 6734void AudioFlinger::EffectModule::process() 6735{ 6736 Mutex::Autolock _l(mLock); 6737 6738 if (mState == DESTROYED || mEffectInterface == NULL || 6739 mConfig.inputCfg.buffer.raw == NULL || 6740 mConfig.outputCfg.buffer.raw == NULL) { 6741 return; 6742 } 6743 6744 if (isProcessEnabled()) { 6745 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6746 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6747 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6748 mConfig.inputCfg.buffer.s32, 6749 mConfig.inputCfg.buffer.frameCount/2); 6750 } 6751 6752 // do the actual processing in the effect engine 6753 int ret = (*mEffectInterface)->process(mEffectInterface, 6754 &mConfig.inputCfg.buffer, 6755 &mConfig.outputCfg.buffer); 6756 6757 // force transition to IDLE state when engine is ready 6758 if (mState == STOPPED && ret == -ENODATA) { 6759 mDisableWaitCnt = 1; 6760 } 6761 6762 // clear auxiliary effect input buffer for next accumulation 6763 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6764 memset(mConfig.inputCfg.buffer.raw, 0, 6765 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6766 } 6767 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6768 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6769 // If an insert effect is idle and input buffer is different from output buffer, 6770 // accumulate input onto output 6771 sp<EffectChain> chain = mChain.promote(); 6772 if (chain != 0 && chain->activeTrackCnt() != 0) { 6773 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6774 int16_t *in = mConfig.inputCfg.buffer.s16; 6775 int16_t *out = mConfig.outputCfg.buffer.s16; 6776 for (size_t i = 0; i < frameCnt; i++) { 6777 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6778 } 6779 } 6780 } 6781} 6782 6783void AudioFlinger::EffectModule::reset_l() 6784{ 6785 if (mEffectInterface == NULL) { 6786 return; 6787 } 6788 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6789} 6790 6791status_t AudioFlinger::EffectModule::configure() 6792{ 6793 uint32_t channels; 6794 if (mEffectInterface == NULL) { 6795 return NO_INIT; 6796 } 6797 6798 sp<ThreadBase> thread = mThread.promote(); 6799 if (thread == 0) { 6800 return DEAD_OBJECT; 6801 } 6802 6803 // TODO: handle configuration of effects replacing track process 6804 if (thread->channelCount() == 1) { 6805 channels = AUDIO_CHANNEL_OUT_MONO; 6806 } else { 6807 channels = AUDIO_CHANNEL_OUT_STEREO; 6808 } 6809 6810 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6811 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6812 } else { 6813 mConfig.inputCfg.channels = channels; 6814 } 6815 mConfig.outputCfg.channels = channels; 6816 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6817 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6818 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6819 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6820 mConfig.inputCfg.bufferProvider.cookie = NULL; 6821 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6822 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6823 mConfig.outputCfg.bufferProvider.cookie = NULL; 6824 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6825 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6826 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6827 // Insert effect: 6828 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6829 // always overwrites output buffer: input buffer == output buffer 6830 // - in other sessions: 6831 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6832 // other effect: overwrites output buffer: input buffer == output buffer 6833 // Auxiliary effect: 6834 // accumulates in output buffer: input buffer != output buffer 6835 // Therefore: accumulate <=> input buffer != output buffer 6836 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6837 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6838 } else { 6839 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6840 } 6841 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6842 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6843 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6844 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6845 6846 ALOGV("configure() %p thread %p buffer %p framecount %d", 6847 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6848 6849 status_t cmdStatus; 6850 uint32_t size = sizeof(int); 6851 status_t status = (*mEffectInterface)->command(mEffectInterface, 6852 EFFECT_CMD_SET_CONFIG, 6853 sizeof(effect_config_t), 6854 &mConfig, 6855 &size, 6856 &cmdStatus); 6857 if (status == 0) { 6858 status = cmdStatus; 6859 } 6860 6861 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6862 (1000 * mConfig.outputCfg.buffer.frameCount); 6863 6864 return status; 6865} 6866 6867status_t AudioFlinger::EffectModule::init() 6868{ 6869 Mutex::Autolock _l(mLock); 6870 if (mEffectInterface == NULL) { 6871 return NO_INIT; 6872 } 6873 status_t cmdStatus; 6874 uint32_t size = sizeof(status_t); 6875 status_t status = (*mEffectInterface)->command(mEffectInterface, 6876 EFFECT_CMD_INIT, 6877 0, 6878 NULL, 6879 &size, 6880 &cmdStatus); 6881 if (status == 0) { 6882 status = cmdStatus; 6883 } 6884 return status; 6885} 6886 6887status_t AudioFlinger::EffectModule::start() 6888{ 6889 Mutex::Autolock _l(mLock); 6890 return start_l(); 6891} 6892 6893status_t AudioFlinger::EffectModule::start_l() 6894{ 6895 if (mEffectInterface == NULL) { 6896 return NO_INIT; 6897 } 6898 status_t cmdStatus; 6899 uint32_t size = sizeof(status_t); 6900 status_t status = (*mEffectInterface)->command(mEffectInterface, 6901 EFFECT_CMD_ENABLE, 6902 0, 6903 NULL, 6904 &size, 6905 &cmdStatus); 6906 if (status == 0) { 6907 status = cmdStatus; 6908 } 6909 if (status == 0 && 6910 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6911 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6912 sp<ThreadBase> thread = mThread.promote(); 6913 if (thread != 0) { 6914 audio_stream_t *stream = thread->stream(); 6915 if (stream != NULL) { 6916 stream->add_audio_effect(stream, mEffectInterface); 6917 } 6918 } 6919 } 6920 return status; 6921} 6922 6923status_t AudioFlinger::EffectModule::stop() 6924{ 6925 Mutex::Autolock _l(mLock); 6926 return stop_l(); 6927} 6928 6929status_t AudioFlinger::EffectModule::stop_l() 6930{ 6931 if (mEffectInterface == NULL) { 6932 return NO_INIT; 6933 } 6934 status_t cmdStatus; 6935 uint32_t size = sizeof(status_t); 6936 status_t status = (*mEffectInterface)->command(mEffectInterface, 6937 EFFECT_CMD_DISABLE, 6938 0, 6939 NULL, 6940 &size, 6941 &cmdStatus); 6942 if (status == 0) { 6943 status = cmdStatus; 6944 } 6945 if (status == 0 && 6946 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6947 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6948 sp<ThreadBase> thread = mThread.promote(); 6949 if (thread != 0) { 6950 audio_stream_t *stream = thread->stream(); 6951 if (stream != NULL) { 6952 stream->remove_audio_effect(stream, mEffectInterface); 6953 } 6954 } 6955 } 6956 return status; 6957} 6958 6959status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6960 uint32_t cmdSize, 6961 void *pCmdData, 6962 uint32_t *replySize, 6963 void *pReplyData) 6964{ 6965 Mutex::Autolock _l(mLock); 6966// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6967 6968 if (mState == DESTROYED || mEffectInterface == NULL) { 6969 return NO_INIT; 6970 } 6971 status_t status = (*mEffectInterface)->command(mEffectInterface, 6972 cmdCode, 6973 cmdSize, 6974 pCmdData, 6975 replySize, 6976 pReplyData); 6977 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6978 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6979 for (size_t i = 1; i < mHandles.size(); i++) { 6980 sp<EffectHandle> h = mHandles[i].promote(); 6981 if (h != 0) { 6982 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6983 } 6984 } 6985 } 6986 return status; 6987} 6988 6989status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6990{ 6991 6992 Mutex::Autolock _l(mLock); 6993 ALOGV("setEnabled %p enabled %d", this, enabled); 6994 6995 if (enabled != isEnabled()) { 6996 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6997 if (enabled && status != NO_ERROR) { 6998 return status; 6999 } 7000 7001 switch (mState) { 7002 // going from disabled to enabled 7003 case IDLE: 7004 mState = STARTING; 7005 break; 7006 case STOPPED: 7007 mState = RESTART; 7008 break; 7009 case STOPPING: 7010 mState = ACTIVE; 7011 break; 7012 7013 // going from enabled to disabled 7014 case RESTART: 7015 mState = STOPPED; 7016 break; 7017 case STARTING: 7018 mState = IDLE; 7019 break; 7020 case ACTIVE: 7021 mState = STOPPING; 7022 break; 7023 case DESTROYED: 7024 return NO_ERROR; // simply ignore as we are being destroyed 7025 } 7026 for (size_t i = 1; i < mHandles.size(); i++) { 7027 sp<EffectHandle> h = mHandles[i].promote(); 7028 if (h != 0) { 7029 h->setEnabled(enabled); 7030 } 7031 } 7032 } 7033 return NO_ERROR; 7034} 7035 7036bool AudioFlinger::EffectModule::isEnabled() const 7037{ 7038 switch (mState) { 7039 case RESTART: 7040 case STARTING: 7041 case ACTIVE: 7042 return true; 7043 case IDLE: 7044 case STOPPING: 7045 case STOPPED: 7046 case DESTROYED: 7047 default: 7048 return false; 7049 } 7050} 7051 7052bool AudioFlinger::EffectModule::isProcessEnabled() const 7053{ 7054 switch (mState) { 7055 case RESTART: 7056 case ACTIVE: 7057 case STOPPING: 7058 case STOPPED: 7059 return true; 7060 case IDLE: 7061 case STARTING: 7062 case DESTROYED: 7063 default: 7064 return false; 7065 } 7066} 7067 7068status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7069{ 7070 Mutex::Autolock _l(mLock); 7071 status_t status = NO_ERROR; 7072 7073 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7074 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7075 if (isProcessEnabled() && 7076 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7077 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7078 status_t cmdStatus; 7079 uint32_t volume[2]; 7080 uint32_t *pVolume = NULL; 7081 uint32_t size = sizeof(volume); 7082 volume[0] = *left; 7083 volume[1] = *right; 7084 if (controller) { 7085 pVolume = volume; 7086 } 7087 status = (*mEffectInterface)->command(mEffectInterface, 7088 EFFECT_CMD_SET_VOLUME, 7089 size, 7090 volume, 7091 &size, 7092 pVolume); 7093 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7094 *left = volume[0]; 7095 *right = volume[1]; 7096 } 7097 } 7098 return status; 7099} 7100 7101status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7102{ 7103 Mutex::Autolock _l(mLock); 7104 status_t status = NO_ERROR; 7105 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7106 // audio pre processing modules on RecordThread can receive both output and 7107 // input device indication in the same call 7108 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7109 if (dev) { 7110 status_t cmdStatus; 7111 uint32_t size = sizeof(status_t); 7112 7113 status = (*mEffectInterface)->command(mEffectInterface, 7114 EFFECT_CMD_SET_DEVICE, 7115 sizeof(uint32_t), 7116 &dev, 7117 &size, 7118 &cmdStatus); 7119 if (status == NO_ERROR) { 7120 status = cmdStatus; 7121 } 7122 } 7123 dev = device & AUDIO_DEVICE_IN_ALL; 7124 if (dev) { 7125 status_t cmdStatus; 7126 uint32_t size = sizeof(status_t); 7127 7128 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7129 EFFECT_CMD_SET_INPUT_DEVICE, 7130 sizeof(uint32_t), 7131 &dev, 7132 &size, 7133 &cmdStatus); 7134 if (status2 == NO_ERROR) { 7135 status2 = cmdStatus; 7136 } 7137 if (status == NO_ERROR) { 7138 status = status2; 7139 } 7140 } 7141 } 7142 return status; 7143} 7144 7145status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7146{ 7147 Mutex::Autolock _l(mLock); 7148 status_t status = NO_ERROR; 7149 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7150 status_t cmdStatus; 7151 uint32_t size = sizeof(status_t); 7152 status = (*mEffectInterface)->command(mEffectInterface, 7153 EFFECT_CMD_SET_AUDIO_MODE, 7154 sizeof(audio_mode_t), 7155 &mode, 7156 &size, 7157 &cmdStatus); 7158 if (status == NO_ERROR) { 7159 status = cmdStatus; 7160 } 7161 } 7162 return status; 7163} 7164 7165void AudioFlinger::EffectModule::setSuspended(bool suspended) 7166{ 7167 Mutex::Autolock _l(mLock); 7168 mSuspended = suspended; 7169} 7170 7171bool AudioFlinger::EffectModule::suspended() const 7172{ 7173 Mutex::Autolock _l(mLock); 7174 return mSuspended; 7175} 7176 7177status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7178{ 7179 const size_t SIZE = 256; 7180 char buffer[SIZE]; 7181 String8 result; 7182 7183 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7184 result.append(buffer); 7185 7186 bool locked = tryLock(mLock); 7187 // failed to lock - AudioFlinger is probably deadlocked 7188 if (!locked) { 7189 result.append("\t\tCould not lock Fx mutex:\n"); 7190 } 7191 7192 result.append("\t\tSession Status State Engine:\n"); 7193 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7194 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7195 result.append(buffer); 7196 7197 result.append("\t\tDescriptor:\n"); 7198 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7199 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7200 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7201 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7202 result.append(buffer); 7203 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7204 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7205 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7206 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7207 result.append(buffer); 7208 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7209 mDescriptor.apiVersion, 7210 mDescriptor.flags); 7211 result.append(buffer); 7212 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7213 mDescriptor.name); 7214 result.append(buffer); 7215 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7216 mDescriptor.implementor); 7217 result.append(buffer); 7218 7219 result.append("\t\t- Input configuration:\n"); 7220 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7221 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7222 (uint32_t)mConfig.inputCfg.buffer.raw, 7223 mConfig.inputCfg.buffer.frameCount, 7224 mConfig.inputCfg.samplingRate, 7225 mConfig.inputCfg.channels, 7226 mConfig.inputCfg.format); 7227 result.append(buffer); 7228 7229 result.append("\t\t- Output configuration:\n"); 7230 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7231 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7232 (uint32_t)mConfig.outputCfg.buffer.raw, 7233 mConfig.outputCfg.buffer.frameCount, 7234 mConfig.outputCfg.samplingRate, 7235 mConfig.outputCfg.channels, 7236 mConfig.outputCfg.format); 7237 result.append(buffer); 7238 7239 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7240 result.append(buffer); 7241 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7242 for (size_t i = 0; i < mHandles.size(); ++i) { 7243 sp<EffectHandle> handle = mHandles[i].promote(); 7244 if (handle != 0) { 7245 handle->dump(buffer, SIZE); 7246 result.append(buffer); 7247 } 7248 } 7249 7250 result.append("\n"); 7251 7252 write(fd, result.string(), result.length()); 7253 7254 if (locked) { 7255 mLock.unlock(); 7256 } 7257 7258 return NO_ERROR; 7259} 7260 7261// ---------------------------------------------------------------------------- 7262// EffectHandle implementation 7263// ---------------------------------------------------------------------------- 7264 7265#undef LOG_TAG 7266#define LOG_TAG "AudioFlinger::EffectHandle" 7267 7268AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7269 const sp<AudioFlinger::Client>& client, 7270 const sp<IEffectClient>& effectClient, 7271 int32_t priority) 7272 : BnEffect(), 7273 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7274 mPriority(priority), mHasControl(false), mEnabled(false) 7275{ 7276 ALOGV("constructor %p", this); 7277 7278 if (client == 0) { 7279 return; 7280 } 7281 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7282 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7283 if (mCblkMemory != 0) { 7284 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7285 7286 if (mCblk != NULL) { 7287 new(mCblk) effect_param_cblk_t(); 7288 mBuffer = (uint8_t *)mCblk + bufOffset; 7289 } 7290 } else { 7291 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7292 return; 7293 } 7294} 7295 7296AudioFlinger::EffectHandle::~EffectHandle() 7297{ 7298 ALOGV("Destructor %p", this); 7299 disconnect(false); 7300 ALOGV("Destructor DONE %p", this); 7301} 7302 7303status_t AudioFlinger::EffectHandle::enable() 7304{ 7305 ALOGV("enable %p", this); 7306 if (!mHasControl) return INVALID_OPERATION; 7307 if (mEffect == 0) return DEAD_OBJECT; 7308 7309 if (mEnabled) { 7310 return NO_ERROR; 7311 } 7312 7313 mEnabled = true; 7314 7315 sp<ThreadBase> thread = mEffect->thread().promote(); 7316 if (thread != 0) { 7317 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7318 } 7319 7320 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7321 if (mEffect->suspended()) { 7322 return NO_ERROR; 7323 } 7324 7325 status_t status = mEffect->setEnabled(true); 7326 if (status != NO_ERROR) { 7327 if (thread != 0) { 7328 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7329 } 7330 mEnabled = false; 7331 } 7332 return status; 7333} 7334 7335status_t AudioFlinger::EffectHandle::disable() 7336{ 7337 ALOGV("disable %p", this); 7338 if (!mHasControl) return INVALID_OPERATION; 7339 if (mEffect == 0) return DEAD_OBJECT; 7340 7341 if (!mEnabled) { 7342 return NO_ERROR; 7343 } 7344 mEnabled = false; 7345 7346 if (mEffect->suspended()) { 7347 return NO_ERROR; 7348 } 7349 7350 status_t status = mEffect->setEnabled(false); 7351 7352 sp<ThreadBase> thread = mEffect->thread().promote(); 7353 if (thread != 0) { 7354 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7355 } 7356 7357 return status; 7358} 7359 7360void AudioFlinger::EffectHandle::disconnect() 7361{ 7362 disconnect(true); 7363} 7364 7365void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7366{ 7367 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7368 if (mEffect == 0) { 7369 return; 7370 } 7371 mEffect->disconnect(this, unpinIfLast); 7372 7373 if (mHasControl && mEnabled) { 7374 sp<ThreadBase> thread = mEffect->thread().promote(); 7375 if (thread != 0) { 7376 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7377 } 7378 } 7379 7380 // release sp on module => module destructor can be called now 7381 mEffect.clear(); 7382 if (mClient != 0) { 7383 if (mCblk != NULL) { 7384 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7385 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7386 } 7387 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7388 // Client destructor must run with AudioFlinger mutex locked 7389 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7390 mClient.clear(); 7391 } 7392} 7393 7394status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7395 uint32_t cmdSize, 7396 void *pCmdData, 7397 uint32_t *replySize, 7398 void *pReplyData) 7399{ 7400// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7401// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7402 7403 // only get parameter command is permitted for applications not controlling the effect 7404 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7405 return INVALID_OPERATION; 7406 } 7407 if (mEffect == 0) return DEAD_OBJECT; 7408 if (mClient == 0) return INVALID_OPERATION; 7409 7410 // handle commands that are not forwarded transparently to effect engine 7411 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7412 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7413 // no risk to block the whole media server process or mixer threads is we are stuck here 7414 Mutex::Autolock _l(mCblk->lock); 7415 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7416 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7417 mCblk->serverIndex = 0; 7418 mCblk->clientIndex = 0; 7419 return BAD_VALUE; 7420 } 7421 status_t status = NO_ERROR; 7422 while (mCblk->serverIndex < mCblk->clientIndex) { 7423 int reply; 7424 uint32_t rsize = sizeof(int); 7425 int *p = (int *)(mBuffer + mCblk->serverIndex); 7426 int size = *p++; 7427 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7428 ALOGW("command(): invalid parameter block size"); 7429 break; 7430 } 7431 effect_param_t *param = (effect_param_t *)p; 7432 if (param->psize == 0 || param->vsize == 0) { 7433 ALOGW("command(): null parameter or value size"); 7434 mCblk->serverIndex += size; 7435 continue; 7436 } 7437 uint32_t psize = sizeof(effect_param_t) + 7438 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7439 param->vsize; 7440 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7441 psize, 7442 p, 7443 &rsize, 7444 &reply); 7445 // stop at first error encountered 7446 if (ret != NO_ERROR) { 7447 status = ret; 7448 *(int *)pReplyData = reply; 7449 break; 7450 } else if (reply != NO_ERROR) { 7451 *(int *)pReplyData = reply; 7452 break; 7453 } 7454 mCblk->serverIndex += size; 7455 } 7456 mCblk->serverIndex = 0; 7457 mCblk->clientIndex = 0; 7458 return status; 7459 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7460 *(int *)pReplyData = NO_ERROR; 7461 return enable(); 7462 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7463 *(int *)pReplyData = NO_ERROR; 7464 return disable(); 7465 } 7466 7467 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7468} 7469 7470void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7471{ 7472 ALOGV("setControl %p control %d", this, hasControl); 7473 7474 mHasControl = hasControl; 7475 mEnabled = enabled; 7476 7477 if (signal && mEffectClient != 0) { 7478 mEffectClient->controlStatusChanged(hasControl); 7479 } 7480} 7481 7482void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7483 uint32_t cmdSize, 7484 void *pCmdData, 7485 uint32_t replySize, 7486 void *pReplyData) 7487{ 7488 if (mEffectClient != 0) { 7489 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7490 } 7491} 7492 7493 7494 7495void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7496{ 7497 if (mEffectClient != 0) { 7498 mEffectClient->enableStatusChanged(enabled); 7499 } 7500} 7501 7502status_t AudioFlinger::EffectHandle::onTransact( 7503 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7504{ 7505 return BnEffect::onTransact(code, data, reply, flags); 7506} 7507 7508 7509void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7510{ 7511 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7512 7513 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7514 (mClient == 0) ? getpid_cached : mClient->pid(), 7515 mPriority, 7516 mHasControl, 7517 !locked, 7518 mCblk ? mCblk->clientIndex : 0, 7519 mCblk ? mCblk->serverIndex : 0 7520 ); 7521 7522 if (locked) { 7523 mCblk->lock.unlock(); 7524 } 7525} 7526 7527#undef LOG_TAG 7528#define LOG_TAG "AudioFlinger::EffectChain" 7529 7530AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7531 int sessionId) 7532 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7533 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7534 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7535{ 7536 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7537 if (thread == NULL) { 7538 return; 7539 } 7540 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7541 thread->frameCount(); 7542} 7543 7544AudioFlinger::EffectChain::~EffectChain() 7545{ 7546 if (mOwnInBuffer) { 7547 delete mInBuffer; 7548 } 7549 7550} 7551 7552// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7553sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7554{ 7555 size_t size = mEffects.size(); 7556 7557 for (size_t i = 0; i < size; i++) { 7558 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7559 return mEffects[i]; 7560 } 7561 } 7562 return 0; 7563} 7564 7565// getEffectFromId_l() must be called with ThreadBase::mLock held 7566sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7567{ 7568 size_t size = mEffects.size(); 7569 7570 for (size_t i = 0; i < size; i++) { 7571 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7572 if (id == 0 || mEffects[i]->id() == id) { 7573 return mEffects[i]; 7574 } 7575 } 7576 return 0; 7577} 7578 7579// getEffectFromType_l() must be called with ThreadBase::mLock held 7580sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7581 const effect_uuid_t *type) 7582{ 7583 size_t size = mEffects.size(); 7584 7585 for (size_t i = 0; i < size; i++) { 7586 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7587 return mEffects[i]; 7588 } 7589 } 7590 return 0; 7591} 7592 7593// Must be called with EffectChain::mLock locked 7594void AudioFlinger::EffectChain::process_l() 7595{ 7596 sp<ThreadBase> thread = mThread.promote(); 7597 if (thread == 0) { 7598 ALOGW("process_l(): cannot promote mixer thread"); 7599 return; 7600 } 7601 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7602 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7603 // always process effects unless no more tracks are on the session and the effect tail 7604 // has been rendered 7605 bool doProcess = true; 7606 if (!isGlobalSession) { 7607 bool tracksOnSession = (trackCnt() != 0); 7608 7609 if (!tracksOnSession && mTailBufferCount == 0) { 7610 doProcess = false; 7611 } 7612 7613 if (activeTrackCnt() == 0) { 7614 // if no track is active and the effect tail has not been rendered, 7615 // the input buffer must be cleared here as the mixer process will not do it 7616 if (tracksOnSession || mTailBufferCount > 0) { 7617 size_t numSamples = thread->frameCount() * thread->channelCount(); 7618 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7619 if (mTailBufferCount > 0) { 7620 mTailBufferCount--; 7621 } 7622 } 7623 } 7624 } 7625 7626 size_t size = mEffects.size(); 7627 if (doProcess) { 7628 for (size_t i = 0; i < size; i++) { 7629 mEffects[i]->process(); 7630 } 7631 } 7632 for (size_t i = 0; i < size; i++) { 7633 mEffects[i]->updateState(); 7634 } 7635} 7636 7637// addEffect_l() must be called with PlaybackThread::mLock held 7638status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7639{ 7640 effect_descriptor_t desc = effect->desc(); 7641 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7642 7643 Mutex::Autolock _l(mLock); 7644 effect->setChain(this); 7645 sp<ThreadBase> thread = mThread.promote(); 7646 if (thread == 0) { 7647 return NO_INIT; 7648 } 7649 effect->setThread(thread); 7650 7651 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7652 // Auxiliary effects are inserted at the beginning of mEffects vector as 7653 // they are processed first and accumulated in chain input buffer 7654 mEffects.insertAt(effect, 0); 7655 7656 // the input buffer for auxiliary effect contains mono samples in 7657 // 32 bit format. This is to avoid saturation in AudoMixer 7658 // accumulation stage. Saturation is done in EffectModule::process() before 7659 // calling the process in effect engine 7660 size_t numSamples = thread->frameCount(); 7661 int32_t *buffer = new int32_t[numSamples]; 7662 memset(buffer, 0, numSamples * sizeof(int32_t)); 7663 effect->setInBuffer((int16_t *)buffer); 7664 // auxiliary effects output samples to chain input buffer for further processing 7665 // by insert effects 7666 effect->setOutBuffer(mInBuffer); 7667 } else { 7668 // Insert effects are inserted at the end of mEffects vector as they are processed 7669 // after track and auxiliary effects. 7670 // Insert effect order as a function of indicated preference: 7671 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7672 // another effect is present 7673 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7674 // last effect claiming first position 7675 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7676 // first effect claiming last position 7677 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7678 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7679 // already present 7680 7681 size_t size = mEffects.size(); 7682 size_t idx_insert = size; 7683 ssize_t idx_insert_first = -1; 7684 ssize_t idx_insert_last = -1; 7685 7686 for (size_t i = 0; i < size; i++) { 7687 effect_descriptor_t d = mEffects[i]->desc(); 7688 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7689 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7690 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7691 // check invalid effect chaining combinations 7692 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7693 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7694 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7695 return INVALID_OPERATION; 7696 } 7697 // remember position of first insert effect and by default 7698 // select this as insert position for new effect 7699 if (idx_insert == size) { 7700 idx_insert = i; 7701 } 7702 // remember position of last insert effect claiming 7703 // first position 7704 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7705 idx_insert_first = i; 7706 } 7707 // remember position of first insert effect claiming 7708 // last position 7709 if (iPref == EFFECT_FLAG_INSERT_LAST && 7710 idx_insert_last == -1) { 7711 idx_insert_last = i; 7712 } 7713 } 7714 } 7715 7716 // modify idx_insert from first position if needed 7717 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7718 if (idx_insert_last != -1) { 7719 idx_insert = idx_insert_last; 7720 } else { 7721 idx_insert = size; 7722 } 7723 } else { 7724 if (idx_insert_first != -1) { 7725 idx_insert = idx_insert_first + 1; 7726 } 7727 } 7728 7729 // always read samples from chain input buffer 7730 effect->setInBuffer(mInBuffer); 7731 7732 // if last effect in the chain, output samples to chain 7733 // output buffer, otherwise to chain input buffer 7734 if (idx_insert == size) { 7735 if (idx_insert != 0) { 7736 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7737 mEffects[idx_insert-1]->configure(); 7738 } 7739 effect->setOutBuffer(mOutBuffer); 7740 } else { 7741 effect->setOutBuffer(mInBuffer); 7742 } 7743 mEffects.insertAt(effect, idx_insert); 7744 7745 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7746 } 7747 effect->configure(); 7748 return NO_ERROR; 7749} 7750 7751// removeEffect_l() must be called with PlaybackThread::mLock held 7752size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7753{ 7754 Mutex::Autolock _l(mLock); 7755 size_t size = mEffects.size(); 7756 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7757 7758 for (size_t i = 0; i < size; i++) { 7759 if (effect == mEffects[i]) { 7760 // calling stop here will remove pre-processing effect from the audio HAL. 7761 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7762 // the middle of a read from audio HAL 7763 if (mEffects[i]->state() == EffectModule::ACTIVE || 7764 mEffects[i]->state() == EffectModule::STOPPING) { 7765 mEffects[i]->stop(); 7766 } 7767 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7768 delete[] effect->inBuffer(); 7769 } else { 7770 if (i == size - 1 && i != 0) { 7771 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7772 mEffects[i - 1]->configure(); 7773 } 7774 } 7775 mEffects.removeAt(i); 7776 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7777 break; 7778 } 7779 } 7780 7781 return mEffects.size(); 7782} 7783 7784// setDevice_l() must be called with PlaybackThread::mLock held 7785void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7786{ 7787 size_t size = mEffects.size(); 7788 for (size_t i = 0; i < size; i++) { 7789 mEffects[i]->setDevice(device); 7790 } 7791} 7792 7793// setMode_l() must be called with PlaybackThread::mLock held 7794void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7795{ 7796 size_t size = mEffects.size(); 7797 for (size_t i = 0; i < size; i++) { 7798 mEffects[i]->setMode(mode); 7799 } 7800} 7801 7802// setVolume_l() must be called with PlaybackThread::mLock held 7803bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7804{ 7805 uint32_t newLeft = *left; 7806 uint32_t newRight = *right; 7807 bool hasControl = false; 7808 int ctrlIdx = -1; 7809 size_t size = mEffects.size(); 7810 7811 // first update volume controller 7812 for (size_t i = size; i > 0; i--) { 7813 if (mEffects[i - 1]->isProcessEnabled() && 7814 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7815 ctrlIdx = i - 1; 7816 hasControl = true; 7817 break; 7818 } 7819 } 7820 7821 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7822 if (hasControl) { 7823 *left = mNewLeftVolume; 7824 *right = mNewRightVolume; 7825 } 7826 return hasControl; 7827 } 7828 7829 mVolumeCtrlIdx = ctrlIdx; 7830 mLeftVolume = newLeft; 7831 mRightVolume = newRight; 7832 7833 // second get volume update from volume controller 7834 if (ctrlIdx >= 0) { 7835 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7836 mNewLeftVolume = newLeft; 7837 mNewRightVolume = newRight; 7838 } 7839 // then indicate volume to all other effects in chain. 7840 // Pass altered volume to effects before volume controller 7841 // and requested volume to effects after controller 7842 uint32_t lVol = newLeft; 7843 uint32_t rVol = newRight; 7844 7845 for (size_t i = 0; i < size; i++) { 7846 if ((int)i == ctrlIdx) continue; 7847 // this also works for ctrlIdx == -1 when there is no volume controller 7848 if ((int)i > ctrlIdx) { 7849 lVol = *left; 7850 rVol = *right; 7851 } 7852 mEffects[i]->setVolume(&lVol, &rVol, false); 7853 } 7854 *left = newLeft; 7855 *right = newRight; 7856 7857 return hasControl; 7858} 7859 7860status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7861{ 7862 const size_t SIZE = 256; 7863 char buffer[SIZE]; 7864 String8 result; 7865 7866 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7867 result.append(buffer); 7868 7869 bool locked = tryLock(mLock); 7870 // failed to lock - AudioFlinger is probably deadlocked 7871 if (!locked) { 7872 result.append("\tCould not lock mutex:\n"); 7873 } 7874 7875 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7876 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7877 mEffects.size(), 7878 (uint32_t)mInBuffer, 7879 (uint32_t)mOutBuffer, 7880 mActiveTrackCnt); 7881 result.append(buffer); 7882 write(fd, result.string(), result.size()); 7883 7884 for (size_t i = 0; i < mEffects.size(); ++i) { 7885 sp<EffectModule> effect = mEffects[i]; 7886 if (effect != 0) { 7887 effect->dump(fd, args); 7888 } 7889 } 7890 7891 if (locked) { 7892 mLock.unlock(); 7893 } 7894 7895 return NO_ERROR; 7896} 7897 7898// must be called with ThreadBase::mLock held 7899void AudioFlinger::EffectChain::setEffectSuspended_l( 7900 const effect_uuid_t *type, bool suspend) 7901{ 7902 sp<SuspendedEffectDesc> desc; 7903 // use effect type UUID timelow as key as there is no real risk of identical 7904 // timeLow fields among effect type UUIDs. 7905 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7906 if (suspend) { 7907 if (index >= 0) { 7908 desc = mSuspendedEffects.valueAt(index); 7909 } else { 7910 desc = new SuspendedEffectDesc(); 7911 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7912 mSuspendedEffects.add(type->timeLow, desc); 7913 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7914 } 7915 if (desc->mRefCount++ == 0) { 7916 sp<EffectModule> effect = getEffectIfEnabled(type); 7917 if (effect != 0) { 7918 desc->mEffect = effect; 7919 effect->setSuspended(true); 7920 effect->setEnabled(false); 7921 } 7922 } 7923 } else { 7924 if (index < 0) { 7925 return; 7926 } 7927 desc = mSuspendedEffects.valueAt(index); 7928 if (desc->mRefCount <= 0) { 7929 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7930 desc->mRefCount = 1; 7931 } 7932 if (--desc->mRefCount == 0) { 7933 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7934 if (desc->mEffect != 0) { 7935 sp<EffectModule> effect = desc->mEffect.promote(); 7936 if (effect != 0) { 7937 effect->setSuspended(false); 7938 sp<EffectHandle> handle = effect->controlHandle(); 7939 if (handle != 0) { 7940 effect->setEnabled(handle->enabled()); 7941 } 7942 } 7943 desc->mEffect.clear(); 7944 } 7945 mSuspendedEffects.removeItemsAt(index); 7946 } 7947 } 7948} 7949 7950// must be called with ThreadBase::mLock held 7951void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7952{ 7953 sp<SuspendedEffectDesc> desc; 7954 7955 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7956 if (suspend) { 7957 if (index >= 0) { 7958 desc = mSuspendedEffects.valueAt(index); 7959 } else { 7960 desc = new SuspendedEffectDesc(); 7961 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7962 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7963 } 7964 if (desc->mRefCount++ == 0) { 7965 Vector< sp<EffectModule> > effects; 7966 getSuspendEligibleEffects(effects); 7967 for (size_t i = 0; i < effects.size(); i++) { 7968 setEffectSuspended_l(&effects[i]->desc().type, true); 7969 } 7970 } 7971 } else { 7972 if (index < 0) { 7973 return; 7974 } 7975 desc = mSuspendedEffects.valueAt(index); 7976 if (desc->mRefCount <= 0) { 7977 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7978 desc->mRefCount = 1; 7979 } 7980 if (--desc->mRefCount == 0) { 7981 Vector<const effect_uuid_t *> types; 7982 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7983 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7984 continue; 7985 } 7986 types.add(&mSuspendedEffects.valueAt(i)->mType); 7987 } 7988 for (size_t i = 0; i < types.size(); i++) { 7989 setEffectSuspended_l(types[i], false); 7990 } 7991 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7992 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7993 } 7994 } 7995} 7996 7997 7998// The volume effect is used for automated tests only 7999#ifndef OPENSL_ES_H_ 8000static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8001 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8002const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8003#endif //OPENSL_ES_H_ 8004 8005bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8006{ 8007 // auxiliary effects and visualizer are never suspended on output mix 8008 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8009 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8010 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8011 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8012 return false; 8013 } 8014 return true; 8015} 8016 8017void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8018{ 8019 effects.clear(); 8020 for (size_t i = 0; i < mEffects.size(); i++) { 8021 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8022 effects.add(mEffects[i]); 8023 } 8024 } 8025} 8026 8027sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8028 const effect_uuid_t *type) 8029{ 8030 sp<EffectModule> effect = getEffectFromType_l(type); 8031 return effect != 0 && effect->isEnabled() ? effect : 0; 8032} 8033 8034void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8035 bool enabled) 8036{ 8037 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8038 if (enabled) { 8039 if (index < 0) { 8040 // if the effect is not suspend check if all effects are suspended 8041 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8042 if (index < 0) { 8043 return; 8044 } 8045 if (!isEffectEligibleForSuspend(effect->desc())) { 8046 return; 8047 } 8048 setEffectSuspended_l(&effect->desc().type, enabled); 8049 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8050 if (index < 0) { 8051 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8052 return; 8053 } 8054 } 8055 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8056 effect->desc().type.timeLow); 8057 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8058 // if effect is requested to suspended but was not yet enabled, supend it now. 8059 if (desc->mEffect == 0) { 8060 desc->mEffect = effect; 8061 effect->setEnabled(false); 8062 effect->setSuspended(true); 8063 } 8064 } else { 8065 if (index < 0) { 8066 return; 8067 } 8068 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8069 effect->desc().type.timeLow); 8070 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8071 desc->mEffect.clear(); 8072 effect->setSuspended(false); 8073 } 8074} 8075 8076#undef LOG_TAG 8077#define LOG_TAG "AudioFlinger" 8078 8079// ---------------------------------------------------------------------------- 8080 8081status_t AudioFlinger::onTransact( 8082 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8083{ 8084 return BnAudioFlinger::onTransact(code, data, reply, flags); 8085} 8086 8087}; // namespace android 8088