AudioFlinger.cpp revision eb3c337a3d6c74ec857dfc8be7eeafe634614bcd
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 "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85 86 87nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 88 89uint32_t AudioFlinger::mScreenState; 90 91#ifdef TEE_SINK 92bool AudioFlinger::mTeeSinkInputEnabled = false; 93bool AudioFlinger::mTeeSinkOutputEnabled = false; 94bool AudioFlinger::mTeeSinkTrackEnabled = false; 95 96size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 97size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 98size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 99#endif 100 101// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 102// we define a minimum time during which a global effect is considered enabled. 103static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 104 105// ---------------------------------------------------------------------------- 106 107static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 108{ 109 const hw_module_t *mod; 110 int rc; 111 112 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 113 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 114 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 115 if (rc) { 116 goto out; 117 } 118 rc = audio_hw_device_open(mod, dev); 119 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 120 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 121 if (rc) { 122 goto out; 123 } 124 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 125 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 126 rc = BAD_VALUE; 127 goto out; 128 } 129 return 0; 130 131out: 132 *dev = NULL; 133 return rc; 134} 135 136// ---------------------------------------------------------------------------- 137 138AudioFlinger::AudioFlinger() 139 : BnAudioFlinger(), 140 mPrimaryHardwareDev(NULL), 141 mHardwareStatus(AUDIO_HW_IDLE), 142 mMasterVolume(1.0f), 143 mMasterMute(false), 144 mNextUniqueId(1), 145 mMode(AUDIO_MODE_INVALID), 146 mBtNrecIsOff(false), 147 mIsLowRamDevice(true), 148 mIsDeviceTypeKnown(false), 149 mGlobalEffectEnableTime(0) 150{ 151 getpid_cached = getpid(); 152 char value[PROPERTY_VALUE_MAX]; 153 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 154 if (doLog) { 155 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters"); 156 } 157#ifdef TEE_SINK 158 (void) property_get("ro.debuggable", value, "0"); 159 int debuggable = atoi(value); 160 int teeEnabled = 0; 161 if (debuggable) { 162 (void) property_get("af.tee", value, "0"); 163 teeEnabled = atoi(value); 164 } 165 if (teeEnabled & 1) 166 mTeeSinkInputEnabled = true; 167 if (teeEnabled & 2) 168 mTeeSinkOutputEnabled = true; 169 if (teeEnabled & 4) 170 mTeeSinkTrackEnabled = true; 171#endif 172} 173 174void AudioFlinger::onFirstRef() 175{ 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 mMode = AUDIO_MODE_NORMAL; 195} 196 197AudioFlinger::~AudioFlinger() 198{ 199 while (!mRecordThreads.isEmpty()) { 200 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 201 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 202 } 203 while (!mPlaybackThreads.isEmpty()) { 204 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 205 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 206 } 207 208 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 209 // no mHardwareLock needed, as there are no other references to this 210 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 211 delete mAudioHwDevs.valueAt(i); 212 } 213} 214 215static const char * const audio_interfaces[] = { 216 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 217 AUDIO_HARDWARE_MODULE_ID_A2DP, 218 AUDIO_HARDWARE_MODULE_ID_USB, 219}; 220#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 221 222AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 223 audio_module_handle_t module, 224 audio_devices_t devices) 225{ 226 // if module is 0, the request comes from an old policy manager and we should load 227 // well known modules 228 if (module == 0) { 229 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 230 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 231 loadHwModule_l(audio_interfaces[i]); 232 } 233 // then try to find a module supporting the requested device. 234 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 235 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 236 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 237 if ((dev->get_supported_devices != NULL) && 238 (dev->get_supported_devices(dev) & devices) == devices) 239 return audioHwDevice; 240 } 241 } else { 242 // check a match for the requested module handle 243 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 244 if (audioHwDevice != NULL) { 245 return audioHwDevice; 246 } 247 } 248 249 return NULL; 250} 251 252void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 253{ 254 const size_t SIZE = 256; 255 char buffer[SIZE]; 256 String8 result; 257 258 result.append("Clients:\n"); 259 for (size_t i = 0; i < mClients.size(); ++i) { 260 sp<Client> client = mClients.valueAt(i).promote(); 261 if (client != 0) { 262 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 263 result.append(buffer); 264 } 265 } 266 267 result.append("Notification Clients:\n"); 268 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 269 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 270 result.append(buffer); 271 } 272 273 result.append("Global session refs:\n"); 274 result.append(" session pid count\n"); 275 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 276 AudioSessionRef *r = mAudioSessionRefs[i]; 277 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 278 result.append(buffer); 279 } 280 write(fd, result.string(), result.size()); 281} 282 283 284void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 285{ 286 const size_t SIZE = 256; 287 char buffer[SIZE]; 288 String8 result; 289 hardware_call_state hardwareStatus = mHardwareStatus; 290 291 snprintf(buffer, SIZE, "Hardware status: %d\n" 292 "Standby Time mSec: %u\n", 293 hardwareStatus, 294 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 295 result.append(buffer); 296 write(fd, result.string(), result.size()); 297} 298 299void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 snprintf(buffer, SIZE, "Permission Denial: " 305 "can't dump AudioFlinger from pid=%d, uid=%d\n", 306 IPCThreadState::self()->getCallingPid(), 307 IPCThreadState::self()->getCallingUid()); 308 result.append(buffer); 309 write(fd, result.string(), result.size()); 310} 311 312bool AudioFlinger::dumpTryLock(Mutex& mutex) 313{ 314 bool locked = false; 315 for (int i = 0; i < kDumpLockRetries; ++i) { 316 if (mutex.tryLock() == NO_ERROR) { 317 locked = true; 318 break; 319 } 320 usleep(kDumpLockSleepUs); 321 } 322 return locked; 323} 324 325status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 326{ 327 if (!dumpAllowed()) { 328 dumpPermissionDenial(fd, args); 329 } else { 330 // get state of hardware lock 331 bool hardwareLocked = dumpTryLock(mHardwareLock); 332 if (!hardwareLocked) { 333 String8 result(kHardwareLockedString); 334 write(fd, result.string(), result.size()); 335 } else { 336 mHardwareLock.unlock(); 337 } 338 339 bool locked = dumpTryLock(mLock); 340 341 // failed to lock - AudioFlinger is probably deadlocked 342 if (!locked) { 343 String8 result(kDeadlockedString); 344 write(fd, result.string(), result.size()); 345 } 346 347 dumpClients(fd, args); 348 dumpInternals(fd, args); 349 350 // dump playback threads 351 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 352 mPlaybackThreads.valueAt(i)->dump(fd, args); 353 } 354 355 // dump record threads 356 for (size_t i = 0; i < mRecordThreads.size(); i++) { 357 mRecordThreads.valueAt(i)->dump(fd, args); 358 } 359 360 // dump all hardware devs 361 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 362 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 363 dev->dump(dev, fd); 364 } 365 366#ifdef TEE_SINK 367 // dump the serially shared record tee sink 368 if (mRecordTeeSource != 0) { 369 dumpTee(fd, mRecordTeeSource); 370 } 371#endif 372 373 if (locked) { 374 mLock.unlock(); 375 } 376 377 // append a copy of media.log here by forwarding fd to it, but don't attempt 378 // to lookup the service if it's not running, as it will block for a second 379 if (mLogMemoryDealer != 0) { 380 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 381 if (binder != 0) { 382 fdprintf(fd, "\nmedia.log:\n"); 383 Vector<String16> args; 384 binder->dump(fd, args); 385 } 386 } 387 } 388 return NO_ERROR; 389} 390 391sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 392{ 393 // If pid is already in the mClients wp<> map, then use that entry 394 // (for which promote() is always != 0), otherwise create a new entry and Client. 395 sp<Client> client = mClients.valueFor(pid).promote(); 396 if (client == 0) { 397 client = new Client(this, pid); 398 mClients.add(pid, client); 399 } 400 401 return client; 402} 403 404sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 405{ 406 if (mLogMemoryDealer == 0) { 407 return new NBLog::Writer(); 408 } 409 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 410 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 411 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 412 if (binder != 0) { 413 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 414 } 415 return writer; 416} 417 418void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 419{ 420 if (writer == 0) { 421 return; 422 } 423 sp<IMemory> iMemory(writer->getIMemory()); 424 if (iMemory == 0) { 425 return; 426 } 427 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 428 if (binder != 0) { 429 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 430 // Now the media.log remote reference to IMemory is gone. 431 // When our last local reference to IMemory also drops to zero, 432 // the IMemory destructor will deallocate the region from mMemoryDealer. 433 } 434} 435 436// IAudioFlinger interface 437 438 439sp<IAudioTrack> AudioFlinger::createTrack( 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 audio_channel_mask_t channelMask, 444 size_t frameCount, 445 IAudioFlinger::track_flags_t *flags, 446 const sp<IMemory>& sharedBuffer, 447 audio_io_handle_t output, 448 pid_t tid, 449 int *sessionId, 450 String8& name, 451 status_t *status) 452{ 453 sp<PlaybackThread::Track> track; 454 sp<TrackHandle> trackHandle; 455 sp<Client> client; 456 status_t lStatus; 457 int lSessionId; 458 459 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 460 // but if someone uses binder directly they could bypass that and cause us to crash 461 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 462 ALOGE("createTrack() invalid stream type %d", streamType); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 468 // and we don't yet support 8.24 or 32-bit PCM 469 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 470 ALOGE("createTrack() invalid format %d", format); 471 lStatus = BAD_VALUE; 472 goto Exit; 473 } 474 475 { 476 Mutex::Autolock _l(mLock); 477 PlaybackThread *thread = checkPlaybackThread_l(output); 478 PlaybackThread *effectThread = NULL; 479 if (thread == NULL) { 480 ALOGE("no playback thread found for output handle %d", output); 481 lStatus = BAD_VALUE; 482 goto Exit; 483 } 484 485 pid_t pid = IPCThreadState::self()->getCallingPid(); 486 client = registerPid_l(pid); 487 488 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 489 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 490 // check if an effect chain with the same session ID is present on another 491 // output thread and move it here. 492 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 493 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 494 if (mPlaybackThreads.keyAt(i) != output) { 495 uint32_t sessions = t->hasAudioSession(*sessionId); 496 if (sessions & PlaybackThread::EFFECT_SESSION) { 497 effectThread = t.get(); 498 break; 499 } 500 } 501 } 502 lSessionId = *sessionId; 503 } else { 504 // if no audio session id is provided, create one here 505 lSessionId = nextUniqueId(); 506 if (sessionId != NULL) { 507 *sessionId = lSessionId; 508 } 509 } 510 ALOGV("createTrack() lSessionId: %d", lSessionId); 511 512 track = thread->createTrack_l(client, streamType, sampleRate, format, 513 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 514 515 // move effect chain to this output thread if an effect on same session was waiting 516 // for a track to be created 517 if (lStatus == NO_ERROR && effectThread != NULL) { 518 Mutex::Autolock _dl(thread->mLock); 519 Mutex::Autolock _sl(effectThread->mLock); 520 moveEffectChain_l(lSessionId, effectThread, thread, true); 521 } 522 523 // Look for sync events awaiting for a session to be used. 524 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 525 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 526 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 527 if (lStatus == NO_ERROR) { 528 (void) track->setSyncEvent(mPendingSyncEvents[i]); 529 } else { 530 mPendingSyncEvents[i]->cancel(); 531 } 532 mPendingSyncEvents.removeAt(i); 533 i--; 534 } 535 } 536 } 537 } 538 if (lStatus == NO_ERROR) { 539 // s for server's pid, n for normal mixer name, f for fast index 540 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 541 track->fastIndex()); 542 trackHandle = new TrackHandle(track); 543 } else { 544 // remove local strong reference to Client before deleting the Track so that the Client 545 // destructor is called by the TrackBase destructor with mLock held 546 client.clear(); 547 track.clear(); 548 } 549 550Exit: 551 if (status != NULL) { 552 *status = lStatus; 553 } 554 return trackHandle; 555} 556 557uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("sampleRate() unknown thread %d", output); 563 return 0; 564 } 565 return thread->sampleRate(); 566} 567 568int AudioFlinger::channelCount(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("channelCount() unknown thread %d", output); 574 return 0; 575 } 576 return thread->channelCount(); 577} 578 579audio_format_t AudioFlinger::format(audio_io_handle_t output) const 580{ 581 Mutex::Autolock _l(mLock); 582 PlaybackThread *thread = checkPlaybackThread_l(output); 583 if (thread == NULL) { 584 ALOGW("format() unknown thread %d", output); 585 return AUDIO_FORMAT_INVALID; 586 } 587 return thread->format(); 588} 589 590size_t AudioFlinger::frameCount(audio_io_handle_t output) const 591{ 592 Mutex::Autolock _l(mLock); 593 PlaybackThread *thread = checkPlaybackThread_l(output); 594 if (thread == NULL) { 595 ALOGW("frameCount() unknown thread %d", output); 596 return 0; 597 } 598 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 599 // should examine all callers and fix them to handle smaller counts 600 return thread->frameCount(); 601} 602 603uint32_t AudioFlinger::latency(audio_io_handle_t output) const 604{ 605 Mutex::Autolock _l(mLock); 606 PlaybackThread *thread = checkPlaybackThread_l(output); 607 if (thread == NULL) { 608 ALOGW("latency(): no playback thread found for output handle %d", output); 609 return 0; 610 } 611 return thread->latency(); 612} 613 614status_t AudioFlinger::setMasterVolume(float value) 615{ 616 status_t ret = initCheck(); 617 if (ret != NO_ERROR) { 618 return ret; 619 } 620 621 // check calling permissions 622 if (!settingsAllowed()) { 623 return PERMISSION_DENIED; 624 } 625 626 Mutex::Autolock _l(mLock); 627 mMasterVolume = value; 628 629 // Set master volume in the HALs which support it. 630 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 631 AutoMutex lock(mHardwareLock); 632 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 633 634 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 635 if (dev->canSetMasterVolume()) { 636 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 637 } 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 // Now set the master volume in each playback thread. Playback threads 642 // assigned to HALs which do not have master volume support will apply 643 // master volume during the mix operation. Threads with HALs which do 644 // support master volume will simply ignore the setting. 645 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 646 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 647 648 return NO_ERROR; 649} 650 651status_t AudioFlinger::setMode(audio_mode_t mode) 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 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 663 ALOGW("Illegal value: setMode(%d)", mode); 664 return BAD_VALUE; 665 } 666 667 { // scope for the lock 668 AutoMutex lock(mHardwareLock); 669 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 670 mHardwareStatus = AUDIO_HW_SET_MODE; 671 ret = dev->set_mode(dev, mode); 672 mHardwareStatus = AUDIO_HW_IDLE; 673 } 674 675 if (NO_ERROR == ret) { 676 Mutex::Autolock _l(mLock); 677 mMode = mode; 678 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 679 mPlaybackThreads.valueAt(i)->setMode(mode); 680 } 681 682 return ret; 683} 684 685status_t AudioFlinger::setMicMute(bool state) 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return ret; 690 } 691 692 // check calling permissions 693 if (!settingsAllowed()) { 694 return PERMISSION_DENIED; 695 } 696 697 AutoMutex lock(mHardwareLock); 698 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 699 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 700 ret = dev->set_mic_mute(dev, state); 701 mHardwareStatus = AUDIO_HW_IDLE; 702 return ret; 703} 704 705bool AudioFlinger::getMicMute() const 706{ 707 status_t ret = initCheck(); 708 if (ret != NO_ERROR) { 709 return false; 710 } 711 712 bool state = AUDIO_MODE_INVALID; 713 AutoMutex lock(mHardwareLock); 714 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 715 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 716 dev->get_mic_mute(dev, &state); 717 mHardwareStatus = AUDIO_HW_IDLE; 718 return state; 719} 720 721status_t AudioFlinger::setMasterMute(bool muted) 722{ 723 status_t ret = initCheck(); 724 if (ret != NO_ERROR) { 725 return ret; 726 } 727 728 // check calling permissions 729 if (!settingsAllowed()) { 730 return PERMISSION_DENIED; 731 } 732 733 Mutex::Autolock _l(mLock); 734 mMasterMute = muted; 735 736 // Set master mute in the HALs which support it. 737 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 738 AutoMutex lock(mHardwareLock); 739 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 740 741 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 742 if (dev->canSetMasterMute()) { 743 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 744 } 745 mHardwareStatus = AUDIO_HW_IDLE; 746 } 747 748 // Now set the master mute in each playback thread. Playback threads 749 // assigned to HALs which do not have master mute support will apply master 750 // mute during the mix operation. Threads with HALs which do support master 751 // mute will simply ignore the setting. 752 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 753 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 754 755 return NO_ERROR; 756} 757 758float AudioFlinger::masterVolume() const 759{ 760 Mutex::Autolock _l(mLock); 761 return masterVolume_l(); 762} 763 764bool AudioFlinger::masterMute() const 765{ 766 Mutex::Autolock _l(mLock); 767 return masterMute_l(); 768} 769 770float AudioFlinger::masterVolume_l() const 771{ 772 return mMasterVolume; 773} 774 775bool AudioFlinger::masterMute_l() const 776{ 777 return mMasterMute; 778} 779 780status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 781 audio_io_handle_t output) 782{ 783 // check calling permissions 784 if (!settingsAllowed()) { 785 return PERMISSION_DENIED; 786 } 787 788 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 789 ALOGE("setStreamVolume() invalid stream %d", stream); 790 return BAD_VALUE; 791 } 792 793 AutoMutex lock(mLock); 794 PlaybackThread *thread = NULL; 795 if (output) { 796 thread = checkPlaybackThread_l(output); 797 if (thread == NULL) { 798 return BAD_VALUE; 799 } 800 } 801 802 mStreamTypes[stream].volume = value; 803 804 if (thread == NULL) { 805 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 806 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 807 } 808 } else { 809 thread->setStreamVolume(stream, value); 810 } 811 812 return NO_ERROR; 813} 814 815status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 816{ 817 // check calling permissions 818 if (!settingsAllowed()) { 819 return PERMISSION_DENIED; 820 } 821 822 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 823 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 824 ALOGE("setStreamMute() invalid stream %d", stream); 825 return BAD_VALUE; 826 } 827 828 AutoMutex lock(mLock); 829 mStreamTypes[stream].mute = muted; 830 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 831 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 832 833 return NO_ERROR; 834} 835 836float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 837{ 838 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 839 return 0.0f; 840 } 841 842 AutoMutex lock(mLock); 843 float volume; 844 if (output) { 845 PlaybackThread *thread = checkPlaybackThread_l(output); 846 if (thread == NULL) { 847 return 0.0f; 848 } 849 volume = thread->streamVolume(stream); 850 } else { 851 volume = streamVolume_l(stream); 852 } 853 854 return volume; 855} 856 857bool AudioFlinger::streamMute(audio_stream_type_t stream) const 858{ 859 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 860 return true; 861 } 862 863 AutoMutex lock(mLock); 864 return streamMute_l(stream); 865} 866 867status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 868{ 869 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 870 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 871 872 // check calling permissions 873 if (!settingsAllowed()) { 874 return PERMISSION_DENIED; 875 } 876 877 // ioHandle == 0 means the parameters are global to the audio hardware interface 878 if (ioHandle == 0) { 879 Mutex::Autolock _l(mLock); 880 status_t final_result = NO_ERROR; 881 { 882 AutoMutex lock(mHardwareLock); 883 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 884 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 885 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 886 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 887 final_result = result ?: final_result; 888 } 889 mHardwareStatus = AUDIO_HW_IDLE; 890 } 891 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 892 AudioParameter param = AudioParameter(keyValuePairs); 893 String8 value; 894 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 895 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 896 if (mBtNrecIsOff != btNrecIsOff) { 897 for (size_t i = 0; i < mRecordThreads.size(); i++) { 898 sp<RecordThread> thread = mRecordThreads.valueAt(i); 899 audio_devices_t device = thread->inDevice(); 900 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 901 // collect all of the thread's session IDs 902 KeyedVector<int, bool> ids = thread->sessionIds(); 903 // suspend effects associated with those session IDs 904 for (size_t j = 0; j < ids.size(); ++j) { 905 int sessionId = ids.keyAt(j); 906 thread->setEffectSuspended(FX_IID_AEC, 907 suspend, 908 sessionId); 909 thread->setEffectSuspended(FX_IID_NS, 910 suspend, 911 sessionId); 912 } 913 } 914 mBtNrecIsOff = btNrecIsOff; 915 } 916 } 917 String8 screenState; 918 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 919 bool isOff = screenState == "off"; 920 if (isOff != (AudioFlinger::mScreenState & 1)) { 921 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 922 } 923 } 924 return final_result; 925 } 926 927 // hold a strong ref on thread in case closeOutput() or closeInput() is called 928 // and the thread is exited once the lock is released 929 sp<ThreadBase> thread; 930 { 931 Mutex::Autolock _l(mLock); 932 thread = checkPlaybackThread_l(ioHandle); 933 if (thread == 0) { 934 thread = checkRecordThread_l(ioHandle); 935 } else if (thread == primaryPlaybackThread_l()) { 936 // indicate output device change to all input threads for pre processing 937 AudioParameter param = AudioParameter(keyValuePairs); 938 int value; 939 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 940 (value != 0)) { 941 for (size_t i = 0; i < mRecordThreads.size(); i++) { 942 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 943 } 944 } 945 } 946 } 947 if (thread != 0) { 948 return thread->setParameters(keyValuePairs); 949 } 950 return BAD_VALUE; 951} 952 953String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 954{ 955 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 956 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 957 958 Mutex::Autolock _l(mLock); 959 960 if (ioHandle == 0) { 961 String8 out_s8; 962 963 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 964 char *s; 965 { 966 AutoMutex lock(mHardwareLock); 967 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 968 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 969 s = dev->get_parameters(dev, keys.string()); 970 mHardwareStatus = AUDIO_HW_IDLE; 971 } 972 out_s8 += String8(s ? s : ""); 973 free(s); 974 } 975 return out_s8; 976 } 977 978 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 979 if (playbackThread != NULL) { 980 return playbackThread->getParameters(keys); 981 } 982 RecordThread *recordThread = checkRecordThread_l(ioHandle); 983 if (recordThread != NULL) { 984 return recordThread->getParameters(keys); 985 } 986 return String8(""); 987} 988 989size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 990 audio_channel_mask_t channelMask) const 991{ 992 status_t ret = initCheck(); 993 if (ret != NO_ERROR) { 994 return 0; 995 } 996 997 AutoMutex lock(mHardwareLock); 998 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 999 struct audio_config config; 1000 memset(&config, 0, sizeof(config)); 1001 config.sample_rate = sampleRate; 1002 config.channel_mask = channelMask; 1003 config.format = format; 1004 1005 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1006 size_t size = dev->get_input_buffer_size(dev, &config); 1007 mHardwareStatus = AUDIO_HW_IDLE; 1008 return size; 1009} 1010 1011unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1012{ 1013 Mutex::Autolock _l(mLock); 1014 1015 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1016 if (recordThread != NULL) { 1017 return recordThread->getInputFramesLost(); 1018 } 1019 return 0; 1020} 1021 1022status_t AudioFlinger::setVoiceVolume(float value) 1023{ 1024 status_t ret = initCheck(); 1025 if (ret != NO_ERROR) { 1026 return ret; 1027 } 1028 1029 // check calling permissions 1030 if (!settingsAllowed()) { 1031 return PERMISSION_DENIED; 1032 } 1033 1034 AutoMutex lock(mHardwareLock); 1035 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1036 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1037 ret = dev->set_voice_volume(dev, value); 1038 mHardwareStatus = AUDIO_HW_IDLE; 1039 1040 return ret; 1041} 1042 1043status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1044 audio_io_handle_t output) const 1045{ 1046 status_t status; 1047 1048 Mutex::Autolock _l(mLock); 1049 1050 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1051 if (playbackThread != NULL) { 1052 return playbackThread->getRenderPosition(halFrames, dspFrames); 1053 } 1054 1055 return BAD_VALUE; 1056} 1057 1058void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1059{ 1060 1061 Mutex::Autolock _l(mLock); 1062 1063 pid_t pid = IPCThreadState::self()->getCallingPid(); 1064 if (mNotificationClients.indexOfKey(pid) < 0) { 1065 sp<NotificationClient> notificationClient = new NotificationClient(this, 1066 client, 1067 pid); 1068 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1069 1070 mNotificationClients.add(pid, notificationClient); 1071 1072 sp<IBinder> binder = client->asBinder(); 1073 binder->linkToDeath(notificationClient); 1074 1075 // the config change is always sent from playback or record threads to avoid deadlock 1076 // with AudioSystem::gLock 1077 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1078 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1079 } 1080 1081 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1082 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1083 } 1084 } 1085} 1086 1087void AudioFlinger::removeNotificationClient(pid_t pid) 1088{ 1089 Mutex::Autolock _l(mLock); 1090 1091 mNotificationClients.removeItem(pid); 1092 1093 ALOGV("%d died, releasing its sessions", pid); 1094 size_t num = mAudioSessionRefs.size(); 1095 bool removed = false; 1096 for (size_t i = 0; i< num; ) { 1097 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1098 ALOGV(" pid %d @ %d", ref->mPid, i); 1099 if (ref->mPid == pid) { 1100 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1101 mAudioSessionRefs.removeAt(i); 1102 delete ref; 1103 removed = true; 1104 num--; 1105 } else { 1106 i++; 1107 } 1108 } 1109 if (removed) { 1110 purgeStaleEffects_l(); 1111 } 1112} 1113 1114// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1115void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1116{ 1117 size_t size = mNotificationClients.size(); 1118 for (size_t i = 0; i < size; i++) { 1119 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1120 param2); 1121 } 1122} 1123 1124// removeClient_l() must be called with AudioFlinger::mLock held 1125void AudioFlinger::removeClient_l(pid_t pid) 1126{ 1127 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1128 IPCThreadState::self()->getCallingPid()); 1129 mClients.removeItem(pid); 1130} 1131 1132// getEffectThread_l() must be called with AudioFlinger::mLock held 1133sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1134{ 1135 sp<PlaybackThread> thread; 1136 1137 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1138 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1139 ALOG_ASSERT(thread == 0); 1140 thread = mPlaybackThreads.valueAt(i); 1141 } 1142 } 1143 1144 return thread; 1145} 1146 1147 1148 1149// ---------------------------------------------------------------------------- 1150 1151AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1152 : RefBase(), 1153 mAudioFlinger(audioFlinger), 1154 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1155 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1156 mPid(pid), 1157 mTimedTrackCount(0) 1158{ 1159 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1160} 1161 1162// Client destructor must be called with AudioFlinger::mLock held 1163AudioFlinger::Client::~Client() 1164{ 1165 mAudioFlinger->removeClient_l(mPid); 1166} 1167 1168sp<MemoryDealer> AudioFlinger::Client::heap() const 1169{ 1170 return mMemoryDealer; 1171} 1172 1173// Reserve one of the limited slots for a timed audio track associated 1174// with this client 1175bool AudioFlinger::Client::reserveTimedTrack() 1176{ 1177 const int kMaxTimedTracksPerClient = 4; 1178 1179 Mutex::Autolock _l(mTimedTrackLock); 1180 1181 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1182 ALOGW("can not create timed track - pid %d has exceeded the limit", 1183 mPid); 1184 return false; 1185 } 1186 1187 mTimedTrackCount++; 1188 return true; 1189} 1190 1191// Release a slot for a timed audio track 1192void AudioFlinger::Client::releaseTimedTrack() 1193{ 1194 Mutex::Autolock _l(mTimedTrackLock); 1195 mTimedTrackCount--; 1196} 1197 1198// ---------------------------------------------------------------------------- 1199 1200AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1201 const sp<IAudioFlingerClient>& client, 1202 pid_t pid) 1203 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1204{ 1205} 1206 1207AudioFlinger::NotificationClient::~NotificationClient() 1208{ 1209} 1210 1211void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1212{ 1213 sp<NotificationClient> keep(this); 1214 mAudioFlinger->removeNotificationClient(mPid); 1215} 1216 1217 1218// ---------------------------------------------------------------------------- 1219 1220static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1221 return audio_is_remote_submix_device(inDevice); 1222} 1223 1224sp<IAudioRecord> AudioFlinger::openRecord( 1225 audio_io_handle_t input, 1226 uint32_t sampleRate, 1227 audio_format_t format, 1228 audio_channel_mask_t channelMask, 1229 size_t frameCount, 1230 IAudioFlinger::track_flags_t *flags, 1231 pid_t tid, 1232 int *sessionId, 1233 status_t *status) 1234{ 1235 sp<RecordThread::RecordTrack> recordTrack; 1236 sp<RecordHandle> recordHandle; 1237 sp<Client> client; 1238 status_t lStatus; 1239 RecordThread *thread; 1240 size_t inFrameCount; 1241 int lSessionId; 1242 1243 // check calling permissions 1244 if (!recordingAllowed()) { 1245 lStatus = PERMISSION_DENIED; 1246 goto Exit; 1247 } 1248 1249 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1250 ALOGE("openRecord() invalid format %d", format); 1251 lStatus = BAD_VALUE; 1252 goto Exit; 1253 } 1254 1255 // add client to list 1256 { // scope for mLock 1257 Mutex::Autolock _l(mLock); 1258 thread = checkRecordThread_l(input); 1259 if (thread == NULL) { 1260 lStatus = BAD_VALUE; 1261 goto Exit; 1262 } 1263 1264 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1265 && !captureAudioOutputAllowed()) { 1266 lStatus = PERMISSION_DENIED; 1267 goto Exit; 1268 } 1269 1270 pid_t pid = IPCThreadState::self()->getCallingPid(); 1271 client = registerPid_l(pid); 1272 1273 // If no audio session id is provided, create one here 1274 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1275 lSessionId = *sessionId; 1276 } else { 1277 lSessionId = nextUniqueId(); 1278 if (sessionId != NULL) { 1279 *sessionId = lSessionId; 1280 } 1281 } 1282 // create new record track. 1283 // The record track uses one track in mHardwareMixerThread by convention. 1284 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1285 frameCount, lSessionId, flags, tid, &lStatus); 1286 } 1287 if (lStatus != NO_ERROR) { 1288 // remove local strong reference to Client before deleting the RecordTrack so that the 1289 // Client destructor is called by the TrackBase destructor with mLock held 1290 client.clear(); 1291 recordTrack.clear(); 1292 goto Exit; 1293 } 1294 1295 // return to handle to client 1296 recordHandle = new RecordHandle(recordTrack); 1297 lStatus = NO_ERROR; 1298 1299Exit: 1300 if (status) { 1301 *status = lStatus; 1302 } 1303 return recordHandle; 1304} 1305 1306 1307 1308// ---------------------------------------------------------------------------- 1309 1310audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1311{ 1312 if (!settingsAllowed()) { 1313 return 0; 1314 } 1315 Mutex::Autolock _l(mLock); 1316 return loadHwModule_l(name); 1317} 1318 1319// loadHwModule_l() must be called with AudioFlinger::mLock held 1320audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1321{ 1322 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1323 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1324 ALOGW("loadHwModule() module %s already loaded", name); 1325 return mAudioHwDevs.keyAt(i); 1326 } 1327 } 1328 1329 audio_hw_device_t *dev; 1330 1331 int rc = load_audio_interface(name, &dev); 1332 if (rc) { 1333 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1334 return 0; 1335 } 1336 1337 mHardwareStatus = AUDIO_HW_INIT; 1338 rc = dev->init_check(dev); 1339 mHardwareStatus = AUDIO_HW_IDLE; 1340 if (rc) { 1341 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1342 return 0; 1343 } 1344 1345 // Check and cache this HAL's level of support for master mute and master 1346 // volume. If this is the first HAL opened, and it supports the get 1347 // methods, use the initial values provided by the HAL as the current 1348 // master mute and volume settings. 1349 1350 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1351 { // scope for auto-lock pattern 1352 AutoMutex lock(mHardwareLock); 1353 1354 if (0 == mAudioHwDevs.size()) { 1355 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1356 if (NULL != dev->get_master_volume) { 1357 float mv; 1358 if (OK == dev->get_master_volume(dev, &mv)) { 1359 mMasterVolume = mv; 1360 } 1361 } 1362 1363 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1364 if (NULL != dev->get_master_mute) { 1365 bool mm; 1366 if (OK == dev->get_master_mute(dev, &mm)) { 1367 mMasterMute = mm; 1368 } 1369 } 1370 } 1371 1372 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1373 if ((NULL != dev->set_master_volume) && 1374 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1375 flags = static_cast<AudioHwDevice::Flags>(flags | 1376 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1377 } 1378 1379 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1380 if ((NULL != dev->set_master_mute) && 1381 (OK == dev->set_master_mute(dev, mMasterMute))) { 1382 flags = static_cast<AudioHwDevice::Flags>(flags | 1383 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1384 } 1385 1386 mHardwareStatus = AUDIO_HW_IDLE; 1387 } 1388 1389 audio_module_handle_t handle = nextUniqueId(); 1390 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1391 1392 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1393 name, dev->common.module->name, dev->common.module->id, handle); 1394 1395 return handle; 1396 1397} 1398 1399// ---------------------------------------------------------------------------- 1400 1401uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1402{ 1403 Mutex::Autolock _l(mLock); 1404 PlaybackThread *thread = primaryPlaybackThread_l(); 1405 return thread != NULL ? thread->sampleRate() : 0; 1406} 1407 1408size_t AudioFlinger::getPrimaryOutputFrameCount() 1409{ 1410 Mutex::Autolock _l(mLock); 1411 PlaybackThread *thread = primaryPlaybackThread_l(); 1412 return thread != NULL ? thread->frameCountHAL() : 0; 1413} 1414 1415// ---------------------------------------------------------------------------- 1416 1417status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1418{ 1419 uid_t uid = IPCThreadState::self()->getCallingUid(); 1420 if (uid != AID_SYSTEM) { 1421 return PERMISSION_DENIED; 1422 } 1423 Mutex::Autolock _l(mLock); 1424 if (mIsDeviceTypeKnown) { 1425 return INVALID_OPERATION; 1426 } 1427 mIsLowRamDevice = isLowRamDevice; 1428 mIsDeviceTypeKnown = true; 1429 return NO_ERROR; 1430} 1431 1432// ---------------------------------------------------------------------------- 1433 1434audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1435 audio_devices_t *pDevices, 1436 uint32_t *pSamplingRate, 1437 audio_format_t *pFormat, 1438 audio_channel_mask_t *pChannelMask, 1439 uint32_t *pLatencyMs, 1440 audio_output_flags_t flags, 1441 const audio_offload_info_t *offloadInfo) 1442{ 1443 PlaybackThread *thread = NULL; 1444 struct audio_config config; 1445 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1446 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1447 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1448 if (offloadInfo) { 1449 config.offload_info = *offloadInfo; 1450 } 1451 1452 audio_stream_out_t *outStream = NULL; 1453 AudioHwDevice *outHwDev; 1454 1455 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1456 module, 1457 (pDevices != NULL) ? *pDevices : 0, 1458 config.sample_rate, 1459 config.format, 1460 config.channel_mask, 1461 flags); 1462 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1463 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1464 1465 if (pDevices == NULL || *pDevices == 0) { 1466 return 0; 1467 } 1468 1469 Mutex::Autolock _l(mLock); 1470 1471 outHwDev = findSuitableHwDev_l(module, *pDevices); 1472 if (outHwDev == NULL) 1473 return 0; 1474 1475 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1476 audio_io_handle_t id = nextUniqueId(); 1477 1478 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1479 1480 status_t status = hwDevHal->open_output_stream(hwDevHal, 1481 id, 1482 *pDevices, 1483 (audio_output_flags_t)flags, 1484 &config, 1485 &outStream); 1486 1487 mHardwareStatus = AUDIO_HW_IDLE; 1488 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1489 "Channels %x, status %d", 1490 outStream, 1491 config.sample_rate, 1492 config.format, 1493 config.channel_mask, 1494 status); 1495 1496 if (status == NO_ERROR && outStream != NULL) { 1497 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1498 1499 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1500 thread = new OffloadThread(this, output, id, *pDevices); 1501 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1502 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1503 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1504 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1505 thread = new DirectOutputThread(this, output, id, *pDevices); 1506 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1507 } else { 1508 thread = new MixerThread(this, output, id, *pDevices); 1509 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1510 } 1511 mPlaybackThreads.add(id, thread); 1512 1513 if (pSamplingRate != NULL) { 1514 *pSamplingRate = config.sample_rate; 1515 } 1516 if (pFormat != NULL) { 1517 *pFormat = config.format; 1518 } 1519 if (pChannelMask != NULL) { 1520 *pChannelMask = config.channel_mask; 1521 } 1522 if (pLatencyMs != NULL) { 1523 *pLatencyMs = thread->latency(); 1524 } 1525 1526 // notify client processes of the new output creation 1527 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1528 1529 // the first primary output opened designates the primary hw device 1530 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1531 ALOGI("Using module %d has the primary audio interface", module); 1532 mPrimaryHardwareDev = outHwDev; 1533 1534 AutoMutex lock(mHardwareLock); 1535 mHardwareStatus = AUDIO_HW_SET_MODE; 1536 hwDevHal->set_mode(hwDevHal, mMode); 1537 mHardwareStatus = AUDIO_HW_IDLE; 1538 } 1539 return id; 1540 } 1541 1542 return 0; 1543} 1544 1545audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1546 audio_io_handle_t output2) 1547{ 1548 Mutex::Autolock _l(mLock); 1549 MixerThread *thread1 = checkMixerThread_l(output1); 1550 MixerThread *thread2 = checkMixerThread_l(output2); 1551 1552 if (thread1 == NULL || thread2 == NULL) { 1553 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1554 output2); 1555 return 0; 1556 } 1557 1558 audio_io_handle_t id = nextUniqueId(); 1559 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1560 thread->addOutputTrack(thread2); 1561 mPlaybackThreads.add(id, thread); 1562 // notify client processes of the new output creation 1563 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1564 return id; 1565} 1566 1567status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1568{ 1569 return closeOutput_nonvirtual(output); 1570} 1571 1572status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1573{ 1574 // keep strong reference on the playback thread so that 1575 // it is not destroyed while exit() is executed 1576 sp<PlaybackThread> thread; 1577 { 1578 Mutex::Autolock _l(mLock); 1579 thread = checkPlaybackThread_l(output); 1580 if (thread == NULL) { 1581 return BAD_VALUE; 1582 } 1583 1584 ALOGV("closeOutput() %d", output); 1585 1586 if (thread->type() == ThreadBase::MIXER) { 1587 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1588 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1589 DuplicatingThread *dupThread = 1590 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1591 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1592 1593 } 1594 } 1595 } 1596 1597 1598 mPlaybackThreads.removeItem(output); 1599 // save all effects to the default thread 1600 if (mPlaybackThreads.size()) { 1601 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1602 if (dstThread != NULL) { 1603 // audioflinger lock is held here so the acquisition order of thread locks does not 1604 // matter 1605 Mutex::Autolock _dl(dstThread->mLock); 1606 Mutex::Autolock _sl(thread->mLock); 1607 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1608 for (size_t i = 0; i < effectChains.size(); i ++) { 1609 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1610 } 1611 } 1612 } 1613 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1614 } 1615 thread->exit(); 1616 // The thread entity (active unit of execution) is no longer running here, 1617 // but the ThreadBase container still exists. 1618 1619 if (thread->type() != ThreadBase::DUPLICATING) { 1620 AudioStreamOut *out = thread->clearOutput(); 1621 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1622 // from now on thread->mOutput is NULL 1623 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1624 delete out; 1625 } 1626 return NO_ERROR; 1627} 1628 1629status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1630{ 1631 Mutex::Autolock _l(mLock); 1632 PlaybackThread *thread = checkPlaybackThread_l(output); 1633 1634 if (thread == NULL) { 1635 return BAD_VALUE; 1636 } 1637 1638 ALOGV("suspendOutput() %d", output); 1639 thread->suspend(); 1640 1641 return NO_ERROR; 1642} 1643 1644status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1645{ 1646 Mutex::Autolock _l(mLock); 1647 PlaybackThread *thread = checkPlaybackThread_l(output); 1648 1649 if (thread == NULL) { 1650 return BAD_VALUE; 1651 } 1652 1653 ALOGV("restoreOutput() %d", output); 1654 1655 thread->restore(); 1656 1657 return NO_ERROR; 1658} 1659 1660audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1661 audio_devices_t *pDevices, 1662 uint32_t *pSamplingRate, 1663 audio_format_t *pFormat, 1664 audio_channel_mask_t *pChannelMask) 1665{ 1666 status_t status; 1667 RecordThread *thread = NULL; 1668 struct audio_config config; 1669 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1670 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1671 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1672 1673 uint32_t reqSamplingRate = config.sample_rate; 1674 audio_format_t reqFormat = config.format; 1675 audio_channel_mask_t reqChannels = config.channel_mask; 1676 audio_stream_in_t *inStream = NULL; 1677 AudioHwDevice *inHwDev; 1678 1679 if (pDevices == NULL || *pDevices == 0) { 1680 return 0; 1681 } 1682 1683 Mutex::Autolock _l(mLock); 1684 1685 inHwDev = findSuitableHwDev_l(module, *pDevices); 1686 if (inHwDev == NULL) 1687 return 0; 1688 1689 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1690 audio_io_handle_t id = nextUniqueId(); 1691 1692 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1693 &inStream); 1694 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1695 "status %d", 1696 inStream, 1697 config.sample_rate, 1698 config.format, 1699 config.channel_mask, 1700 status); 1701 1702 // If the input could not be opened with the requested parameters and we can handle the 1703 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1704 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1705 if (status == BAD_VALUE && 1706 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1707 (config.sample_rate <= 2 * reqSamplingRate) && 1708 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1709 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1710 inStream = NULL; 1711 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1712 } 1713 1714 if (status == NO_ERROR && inStream != NULL) { 1715 1716#ifdef TEE_SINK 1717 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1718 // or (re-)create if current Pipe is idle and does not match the new format 1719 sp<NBAIO_Sink> teeSink; 1720 enum { 1721 TEE_SINK_NO, // don't copy input 1722 TEE_SINK_NEW, // copy input using a new pipe 1723 TEE_SINK_OLD, // copy input using an existing pipe 1724 } kind; 1725 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1726 popcount(inStream->common.get_channels(&inStream->common))); 1727 if (!mTeeSinkInputEnabled) { 1728 kind = TEE_SINK_NO; 1729 } else if (format == Format_Invalid) { 1730 kind = TEE_SINK_NO; 1731 } else if (mRecordTeeSink == 0) { 1732 kind = TEE_SINK_NEW; 1733 } else if (mRecordTeeSink->getStrongCount() != 1) { 1734 kind = TEE_SINK_NO; 1735 } else if (format == mRecordTeeSink->format()) { 1736 kind = TEE_SINK_OLD; 1737 } else { 1738 kind = TEE_SINK_NEW; 1739 } 1740 switch (kind) { 1741 case TEE_SINK_NEW: { 1742 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1743 size_t numCounterOffers = 0; 1744 const NBAIO_Format offers[1] = {format}; 1745 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1746 ALOG_ASSERT(index == 0); 1747 PipeReader *pipeReader = new PipeReader(*pipe); 1748 numCounterOffers = 0; 1749 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1750 ALOG_ASSERT(index == 0); 1751 mRecordTeeSink = pipe; 1752 mRecordTeeSource = pipeReader; 1753 teeSink = pipe; 1754 } 1755 break; 1756 case TEE_SINK_OLD: 1757 teeSink = mRecordTeeSink; 1758 break; 1759 case TEE_SINK_NO: 1760 default: 1761 break; 1762 } 1763#endif 1764 1765 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1766 1767 // Start record thread 1768 // RecordThread requires both input and output device indication to forward to audio 1769 // pre processing modules 1770 thread = new RecordThread(this, 1771 input, 1772 reqSamplingRate, 1773 reqChannels, 1774 id, 1775 primaryOutputDevice_l(), 1776 *pDevices 1777#ifdef TEE_SINK 1778 , teeSink 1779#endif 1780 ); 1781 mRecordThreads.add(id, thread); 1782 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1783 if (pSamplingRate != NULL) { 1784 *pSamplingRate = reqSamplingRate; 1785 } 1786 if (pFormat != NULL) { 1787 *pFormat = config.format; 1788 } 1789 if (pChannelMask != NULL) { 1790 *pChannelMask = reqChannels; 1791 } 1792 1793 // notify client processes of the new input creation 1794 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1795 return id; 1796 } 1797 1798 return 0; 1799} 1800 1801status_t AudioFlinger::closeInput(audio_io_handle_t input) 1802{ 1803 return closeInput_nonvirtual(input); 1804} 1805 1806status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1807{ 1808 // keep strong reference on the record thread so that 1809 // it is not destroyed while exit() is executed 1810 sp<RecordThread> thread; 1811 { 1812 Mutex::Autolock _l(mLock); 1813 thread = checkRecordThread_l(input); 1814 if (thread == 0) { 1815 return BAD_VALUE; 1816 } 1817 1818 ALOGV("closeInput() %d", input); 1819 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1820 mRecordThreads.removeItem(input); 1821 } 1822 thread->exit(); 1823 // The thread entity (active unit of execution) is no longer running here, 1824 // but the ThreadBase container still exists. 1825 1826 AudioStreamIn *in = thread->clearInput(); 1827 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1828 // from now on thread->mInput is NULL 1829 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1830 delete in; 1831 1832 return NO_ERROR; 1833} 1834 1835status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1839 1840 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1841 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1842 thread->invalidateTracks(stream); 1843 } 1844 1845 return NO_ERROR; 1846} 1847 1848 1849int AudioFlinger::newAudioSessionId() 1850{ 1851 return nextUniqueId(); 1852} 1853 1854void AudioFlinger::acquireAudioSessionId(int audioSession) 1855{ 1856 Mutex::Autolock _l(mLock); 1857 pid_t caller = IPCThreadState::self()->getCallingPid(); 1858 ALOGV("acquiring %d from %d", audioSession, caller); 1859 1860 // Ignore requests received from processes not known as notification client. The request 1861 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 1862 // called from a different pid leaving a stale session reference. Also we don't know how 1863 // to clear this reference if the client process dies. 1864 if (mNotificationClients.indexOfKey(caller) < 0) { 1865 ALOGV("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 1866 return; 1867 } 1868 1869 size_t num = mAudioSessionRefs.size(); 1870 for (size_t i = 0; i< num; i++) { 1871 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1872 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1873 ref->mCnt++; 1874 ALOGV(" incremented refcount to %d", ref->mCnt); 1875 return; 1876 } 1877 } 1878 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1879 ALOGV(" added new entry for %d", audioSession); 1880} 1881 1882void AudioFlinger::releaseAudioSessionId(int audioSession) 1883{ 1884 Mutex::Autolock _l(mLock); 1885 pid_t caller = IPCThreadState::self()->getCallingPid(); 1886 ALOGV("releasing %d from %d", audioSession, caller); 1887 size_t num = mAudioSessionRefs.size(); 1888 for (size_t i = 0; i< num; i++) { 1889 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1890 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1891 ref->mCnt--; 1892 ALOGV(" decremented refcount to %d", ref->mCnt); 1893 if (ref->mCnt == 0) { 1894 mAudioSessionRefs.removeAt(i); 1895 delete ref; 1896 purgeStaleEffects_l(); 1897 } 1898 return; 1899 } 1900 } 1901 // If the caller is mediaserver it is likely that the session being released was acquired 1902 // on behalf of a process not in notification clients and we ignore the warning. 1903 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 1904} 1905 1906void AudioFlinger::purgeStaleEffects_l() { 1907 1908 ALOGV("purging stale effects"); 1909 1910 Vector< sp<EffectChain> > chains; 1911 1912 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1913 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1914 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1915 sp<EffectChain> ec = t->mEffectChains[j]; 1916 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1917 chains.push(ec); 1918 } 1919 } 1920 } 1921 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1922 sp<RecordThread> t = mRecordThreads.valueAt(i); 1923 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1924 sp<EffectChain> ec = t->mEffectChains[j]; 1925 chains.push(ec); 1926 } 1927 } 1928 1929 for (size_t i = 0; i < chains.size(); i++) { 1930 sp<EffectChain> ec = chains[i]; 1931 int sessionid = ec->sessionId(); 1932 sp<ThreadBase> t = ec->mThread.promote(); 1933 if (t == 0) { 1934 continue; 1935 } 1936 size_t numsessionrefs = mAudioSessionRefs.size(); 1937 bool found = false; 1938 for (size_t k = 0; k < numsessionrefs; k++) { 1939 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1940 if (ref->mSessionid == sessionid) { 1941 ALOGV(" session %d still exists for %d with %d refs", 1942 sessionid, ref->mPid, ref->mCnt); 1943 found = true; 1944 break; 1945 } 1946 } 1947 if (!found) { 1948 Mutex::Autolock _l (t->mLock); 1949 // remove all effects from the chain 1950 while (ec->mEffects.size()) { 1951 sp<EffectModule> effect = ec->mEffects[0]; 1952 effect->unPin(); 1953 t->removeEffect_l(effect); 1954 if (effect->purgeHandles()) { 1955 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1956 } 1957 AudioSystem::unregisterEffect(effect->id()); 1958 } 1959 } 1960 } 1961 return; 1962} 1963 1964// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1965AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1966{ 1967 return mPlaybackThreads.valueFor(output).get(); 1968} 1969 1970// checkMixerThread_l() must be called with AudioFlinger::mLock held 1971AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1972{ 1973 PlaybackThread *thread = checkPlaybackThread_l(output); 1974 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1975} 1976 1977// checkRecordThread_l() must be called with AudioFlinger::mLock held 1978AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1979{ 1980 return mRecordThreads.valueFor(input).get(); 1981} 1982 1983uint32_t AudioFlinger::nextUniqueId() 1984{ 1985 return android_atomic_inc(&mNextUniqueId); 1986} 1987 1988AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1989{ 1990 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1991 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1992 AudioStreamOut *output = thread->getOutput(); 1993 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1994 return thread; 1995 } 1996 } 1997 return NULL; 1998} 1999 2000audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2001{ 2002 PlaybackThread *thread = primaryPlaybackThread_l(); 2003 2004 if (thread == NULL) { 2005 return 0; 2006 } 2007 2008 return thread->outDevice(); 2009} 2010 2011sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2012 int triggerSession, 2013 int listenerSession, 2014 sync_event_callback_t callBack, 2015 void *cookie) 2016{ 2017 Mutex::Autolock _l(mLock); 2018 2019 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2020 status_t playStatus = NAME_NOT_FOUND; 2021 status_t recStatus = NAME_NOT_FOUND; 2022 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2023 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2024 if (playStatus == NO_ERROR) { 2025 return event; 2026 } 2027 } 2028 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2029 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2030 if (recStatus == NO_ERROR) { 2031 return event; 2032 } 2033 } 2034 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2035 mPendingSyncEvents.add(event); 2036 } else { 2037 ALOGV("createSyncEvent() invalid event %d", event->type()); 2038 event.clear(); 2039 } 2040 return event; 2041} 2042 2043// ---------------------------------------------------------------------------- 2044// Effect management 2045// ---------------------------------------------------------------------------- 2046 2047 2048status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2049{ 2050 Mutex::Autolock _l(mLock); 2051 return EffectQueryNumberEffects(numEffects); 2052} 2053 2054status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2055{ 2056 Mutex::Autolock _l(mLock); 2057 return EffectQueryEffect(index, descriptor); 2058} 2059 2060status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2061 effect_descriptor_t *descriptor) const 2062{ 2063 Mutex::Autolock _l(mLock); 2064 return EffectGetDescriptor(pUuid, descriptor); 2065} 2066 2067 2068sp<IEffect> AudioFlinger::createEffect( 2069 effect_descriptor_t *pDesc, 2070 const sp<IEffectClient>& effectClient, 2071 int32_t priority, 2072 audio_io_handle_t io, 2073 int sessionId, 2074 status_t *status, 2075 int *id, 2076 int *enabled) 2077{ 2078 status_t lStatus = NO_ERROR; 2079 sp<EffectHandle> handle; 2080 effect_descriptor_t desc; 2081 2082 pid_t pid = IPCThreadState::self()->getCallingPid(); 2083 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2084 pid, effectClient.get(), priority, sessionId, io); 2085 2086 if (pDesc == NULL) { 2087 lStatus = BAD_VALUE; 2088 goto Exit; 2089 } 2090 2091 // check audio settings permission for global effects 2092 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2093 lStatus = PERMISSION_DENIED; 2094 goto Exit; 2095 } 2096 2097 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2098 // that can only be created by audio policy manager (running in same process) 2099 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2100 lStatus = PERMISSION_DENIED; 2101 goto Exit; 2102 } 2103 2104 { 2105 if (!EffectIsNullUuid(&pDesc->uuid)) { 2106 // if uuid is specified, request effect descriptor 2107 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2108 if (lStatus < 0) { 2109 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2110 goto Exit; 2111 } 2112 } else { 2113 // if uuid is not specified, look for an available implementation 2114 // of the required type in effect factory 2115 if (EffectIsNullUuid(&pDesc->type)) { 2116 ALOGW("createEffect() no effect type"); 2117 lStatus = BAD_VALUE; 2118 goto Exit; 2119 } 2120 uint32_t numEffects = 0; 2121 effect_descriptor_t d; 2122 d.flags = 0; // prevent compiler warning 2123 bool found = false; 2124 2125 lStatus = EffectQueryNumberEffects(&numEffects); 2126 if (lStatus < 0) { 2127 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2128 goto Exit; 2129 } 2130 for (uint32_t i = 0; i < numEffects; i++) { 2131 lStatus = EffectQueryEffect(i, &desc); 2132 if (lStatus < 0) { 2133 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2134 continue; 2135 } 2136 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2137 // If matching type found save effect descriptor. If the session is 2138 // 0 and the effect is not auxiliary, continue enumeration in case 2139 // an auxiliary version of this effect type is available 2140 found = true; 2141 d = desc; 2142 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2143 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2144 break; 2145 } 2146 } 2147 } 2148 if (!found) { 2149 lStatus = BAD_VALUE; 2150 ALOGW("createEffect() effect not found"); 2151 goto Exit; 2152 } 2153 // For same effect type, chose auxiliary version over insert version if 2154 // connect to output mix (Compliance to OpenSL ES) 2155 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2156 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2157 desc = d; 2158 } 2159 } 2160 2161 // Do not allow auxiliary effects on a session different from 0 (output mix) 2162 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2163 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2164 lStatus = INVALID_OPERATION; 2165 goto Exit; 2166 } 2167 2168 // check recording permission for visualizer 2169 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2170 !recordingAllowed()) { 2171 lStatus = PERMISSION_DENIED; 2172 goto Exit; 2173 } 2174 2175 // return effect descriptor 2176 *pDesc = desc; 2177 if (io == 0 && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2178 // if the output returned by getOutputForEffect() is removed before we lock the 2179 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2180 // and we will exit safely 2181 io = AudioSystem::getOutputForEffect(&desc); 2182 ALOGV("createEffect got output %d", io); 2183 } 2184 2185 Mutex::Autolock _l(mLock); 2186 2187 // If output is not specified try to find a matching audio session ID in one of the 2188 // output threads. 2189 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2190 // because of code checking output when entering the function. 2191 // Note: io is never 0 when creating an effect on an input 2192 if (io == 0) { 2193 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2194 // output must be specified by AudioPolicyManager when using session 2195 // AUDIO_SESSION_OUTPUT_STAGE 2196 lStatus = BAD_VALUE; 2197 goto Exit; 2198 } 2199 // look for the thread where the specified audio session is present 2200 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2201 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2202 io = mPlaybackThreads.keyAt(i); 2203 break; 2204 } 2205 } 2206 if (io == 0) { 2207 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2208 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2209 io = mRecordThreads.keyAt(i); 2210 break; 2211 } 2212 } 2213 } 2214 // If no output thread contains the requested session ID, default to 2215 // first output. The effect chain will be moved to the correct output 2216 // thread when a track with the same session ID is created 2217 if (io == 0 && mPlaybackThreads.size()) { 2218 io = mPlaybackThreads.keyAt(0); 2219 } 2220 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2221 } 2222 ThreadBase *thread = checkRecordThread_l(io); 2223 if (thread == NULL) { 2224 thread = checkPlaybackThread_l(io); 2225 if (thread == NULL) { 2226 ALOGE("createEffect() unknown output thread"); 2227 lStatus = BAD_VALUE; 2228 goto Exit; 2229 } 2230 } 2231 2232 sp<Client> client = registerPid_l(pid); 2233 2234 // create effect on selected output thread 2235 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2236 &desc, enabled, &lStatus); 2237 if (handle != 0 && id != NULL) { 2238 *id = handle->id(); 2239 } 2240 } 2241 2242Exit: 2243 if (status != NULL) { 2244 *status = lStatus; 2245 } 2246 return handle; 2247} 2248 2249status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2250 audio_io_handle_t dstOutput) 2251{ 2252 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2253 sessionId, srcOutput, dstOutput); 2254 Mutex::Autolock _l(mLock); 2255 if (srcOutput == dstOutput) { 2256 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2257 return NO_ERROR; 2258 } 2259 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2260 if (srcThread == NULL) { 2261 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2262 return BAD_VALUE; 2263 } 2264 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2265 if (dstThread == NULL) { 2266 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2267 return BAD_VALUE; 2268 } 2269 2270 Mutex::Autolock _dl(dstThread->mLock); 2271 Mutex::Autolock _sl(srcThread->mLock); 2272 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2273} 2274 2275// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2276status_t AudioFlinger::moveEffectChain_l(int sessionId, 2277 AudioFlinger::PlaybackThread *srcThread, 2278 AudioFlinger::PlaybackThread *dstThread, 2279 bool reRegister) 2280{ 2281 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2282 sessionId, srcThread, dstThread); 2283 2284 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2285 if (chain == 0) { 2286 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2287 sessionId, srcThread); 2288 return INVALID_OPERATION; 2289 } 2290 2291 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2292 // so that a new chain is created with correct parameters when first effect is added. This is 2293 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2294 // removed. 2295 srcThread->removeEffectChain_l(chain); 2296 2297 // transfer all effects one by one so that new effect chain is created on new thread with 2298 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2299 sp<EffectChain> dstChain; 2300 uint32_t strategy = 0; // prevent compiler warning 2301 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2302 Vector< sp<EffectModule> > removed; 2303 status_t status = NO_ERROR; 2304 while (effect != 0) { 2305 srcThread->removeEffect_l(effect); 2306 removed.add(effect); 2307 status = dstThread->addEffect_l(effect); 2308 if (status != NO_ERROR) { 2309 break; 2310 } 2311 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2312 if (effect->state() == EffectModule::ACTIVE || 2313 effect->state() == EffectModule::STOPPING) { 2314 effect->start(); 2315 } 2316 // if the move request is not received from audio policy manager, the effect must be 2317 // re-registered with the new strategy and output 2318 if (dstChain == 0) { 2319 dstChain = effect->chain().promote(); 2320 if (dstChain == 0) { 2321 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2322 status = NO_INIT; 2323 break; 2324 } 2325 strategy = dstChain->strategy(); 2326 } 2327 if (reRegister) { 2328 AudioSystem::unregisterEffect(effect->id()); 2329 AudioSystem::registerEffect(&effect->desc(), 2330 dstThread->id(), 2331 strategy, 2332 sessionId, 2333 effect->id()); 2334 } 2335 effect = chain->getEffectFromId_l(0); 2336 } 2337 2338 if (status != NO_ERROR) { 2339 for (size_t i = 0; i < removed.size(); i++) { 2340 srcThread->addEffect_l(removed[i]); 2341 if (dstChain != 0 && reRegister) { 2342 AudioSystem::unregisterEffect(removed[i]->id()); 2343 AudioSystem::registerEffect(&removed[i]->desc(), 2344 srcThread->id(), 2345 strategy, 2346 sessionId, 2347 removed[i]->id()); 2348 } 2349 } 2350 } 2351 2352 return status; 2353} 2354 2355bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2356{ 2357 if (mGlobalEffectEnableTime != 0 && 2358 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2359 return true; 2360 } 2361 2362 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2363 sp<EffectChain> ec = 2364 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2365 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2366 return true; 2367 } 2368 } 2369 return false; 2370} 2371 2372void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2373{ 2374 Mutex::Autolock _l(mLock); 2375 2376 mGlobalEffectEnableTime = systemTime(); 2377 2378 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2379 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2380 if (t->mType == ThreadBase::OFFLOAD) { 2381 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2382 } 2383 } 2384 2385} 2386 2387struct Entry { 2388#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2389 char mName[MAX_NAME]; 2390}; 2391 2392int comparEntry(const void *p1, const void *p2) 2393{ 2394 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2395} 2396 2397#ifdef TEE_SINK 2398void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2399{ 2400 NBAIO_Source *teeSource = source.get(); 2401 if (teeSource != NULL) { 2402 // .wav rotation 2403 // There is a benign race condition if 2 threads call this simultaneously. 2404 // They would both traverse the directory, but the result would simply be 2405 // failures at unlink() which are ignored. It's also unlikely since 2406 // normally dumpsys is only done by bugreport or from the command line. 2407 char teePath[32+256]; 2408 strcpy(teePath, "/data/misc/media"); 2409 size_t teePathLen = strlen(teePath); 2410 DIR *dir = opendir(teePath); 2411 teePath[teePathLen++] = '/'; 2412 if (dir != NULL) { 2413#define MAX_SORT 20 // number of entries to sort 2414#define MAX_KEEP 10 // number of entries to keep 2415 struct Entry entries[MAX_SORT]; 2416 size_t entryCount = 0; 2417 while (entryCount < MAX_SORT) { 2418 struct dirent de; 2419 struct dirent *result = NULL; 2420 int rc = readdir_r(dir, &de, &result); 2421 if (rc != 0) { 2422 ALOGW("readdir_r failed %d", rc); 2423 break; 2424 } 2425 if (result == NULL) { 2426 break; 2427 } 2428 if (result != &de) { 2429 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2430 break; 2431 } 2432 // ignore non .wav file entries 2433 size_t nameLen = strlen(de.d_name); 2434 if (nameLen <= 4 || nameLen >= MAX_NAME || 2435 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2436 continue; 2437 } 2438 strcpy(entries[entryCount++].mName, de.d_name); 2439 } 2440 (void) closedir(dir); 2441 if (entryCount > MAX_KEEP) { 2442 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2443 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2444 strcpy(&teePath[teePathLen], entries[i].mName); 2445 (void) unlink(teePath); 2446 } 2447 } 2448 } else { 2449 if (fd >= 0) { 2450 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2451 } 2452 } 2453 char teeTime[16]; 2454 struct timeval tv; 2455 gettimeofday(&tv, NULL); 2456 struct tm tm; 2457 localtime_r(&tv.tv_sec, &tm); 2458 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2459 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2460 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2461 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2462 if (teeFd >= 0) { 2463 char wavHeader[44]; 2464 memcpy(wavHeader, 2465 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 2466 sizeof(wavHeader)); 2467 NBAIO_Format format = teeSource->format(); 2468 unsigned channelCount = Format_channelCount(format); 2469 ALOG_ASSERT(channelCount <= FCC_2); 2470 uint32_t sampleRate = Format_sampleRate(format); 2471 wavHeader[22] = channelCount; // number of channels 2472 wavHeader[24] = sampleRate; // sample rate 2473 wavHeader[25] = sampleRate >> 8; 2474 wavHeader[32] = channelCount * 2; // block alignment 2475 write(teeFd, wavHeader, sizeof(wavHeader)); 2476 size_t total = 0; 2477 bool firstRead = true; 2478 for (;;) { 2479#define TEE_SINK_READ 1024 2480 short buffer[TEE_SINK_READ * FCC_2]; 2481 size_t count = TEE_SINK_READ; 2482 ssize_t actual = teeSource->read(buffer, count, 2483 AudioBufferProvider::kInvalidPTS); 2484 bool wasFirstRead = firstRead; 2485 firstRead = false; 2486 if (actual <= 0) { 2487 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2488 continue; 2489 } 2490 break; 2491 } 2492 ALOG_ASSERT(actual <= (ssize_t)count); 2493 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2494 total += actual; 2495 } 2496 lseek(teeFd, (off_t) 4, SEEK_SET); 2497 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2498 write(teeFd, &temp, sizeof(temp)); 2499 lseek(teeFd, (off_t) 40, SEEK_SET); 2500 temp = total * channelCount * sizeof(short); 2501 write(teeFd, &temp, sizeof(temp)); 2502 close(teeFd); 2503 if (fd >= 0) { 2504 fdprintf(fd, "tee copied to %s\n", teePath); 2505 } 2506 } else { 2507 if (fd >= 0) { 2508 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2509 } 2510 } 2511 } 2512} 2513#endif 2514 2515// ---------------------------------------------------------------------------- 2516 2517status_t AudioFlinger::onTransact( 2518 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2519{ 2520 return BnAudioFlinger::onTransact(code, data, reply, flags); 2521} 2522 2523}; // namespace android 2524