AudioFlinger.cpp revision 5baf2af52cd186633b7173196c1e4a4cd3435f22
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("Global session refs:\n"); 268 result.append(" session pid count\n"); 269 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 270 AudioSessionRef *r = mAudioSessionRefs[i]; 271 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 272 result.append(buffer); 273 } 274 write(fd, result.string(), result.size()); 275} 276 277 278void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 279{ 280 const size_t SIZE = 256; 281 char buffer[SIZE]; 282 String8 result; 283 hardware_call_state hardwareStatus = mHardwareStatus; 284 285 snprintf(buffer, SIZE, "Hardware status: %d\n" 286 "Standby Time mSec: %u\n", 287 hardwareStatus, 288 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 289 result.append(buffer); 290 write(fd, result.string(), result.size()); 291} 292 293void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 294{ 295 const size_t SIZE = 256; 296 char buffer[SIZE]; 297 String8 result; 298 snprintf(buffer, SIZE, "Permission Denial: " 299 "can't dump AudioFlinger from pid=%d, uid=%d\n", 300 IPCThreadState::self()->getCallingPid(), 301 IPCThreadState::self()->getCallingUid()); 302 result.append(buffer); 303 write(fd, result.string(), result.size()); 304} 305 306bool AudioFlinger::dumpTryLock(Mutex& mutex) 307{ 308 bool locked = false; 309 for (int i = 0; i < kDumpLockRetries; ++i) { 310 if (mutex.tryLock() == NO_ERROR) { 311 locked = true; 312 break; 313 } 314 usleep(kDumpLockSleepUs); 315 } 316 return locked; 317} 318 319status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 320{ 321 if (!dumpAllowed()) { 322 dumpPermissionDenial(fd, args); 323 } else { 324 // get state of hardware lock 325 bool hardwareLocked = dumpTryLock(mHardwareLock); 326 if (!hardwareLocked) { 327 String8 result(kHardwareLockedString); 328 write(fd, result.string(), result.size()); 329 } else { 330 mHardwareLock.unlock(); 331 } 332 333 bool locked = dumpTryLock(mLock); 334 335 // failed to lock - AudioFlinger is probably deadlocked 336 if (!locked) { 337 String8 result(kDeadlockedString); 338 write(fd, result.string(), result.size()); 339 } 340 341 dumpClients(fd, args); 342 dumpInternals(fd, args); 343 344 // dump playback threads 345 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 346 mPlaybackThreads.valueAt(i)->dump(fd, args); 347 } 348 349 // dump record threads 350 for (size_t i = 0; i < mRecordThreads.size(); i++) { 351 mRecordThreads.valueAt(i)->dump(fd, args); 352 } 353 354 // dump all hardware devs 355 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 356 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 357 dev->dump(dev, fd); 358 } 359 360#ifdef TEE_SINK 361 // dump the serially shared record tee sink 362 if (mRecordTeeSource != 0) { 363 dumpTee(fd, mRecordTeeSource); 364 } 365#endif 366 367 if (locked) { 368 mLock.unlock(); 369 } 370 371 // append a copy of media.log here by forwarding fd to it, but don't attempt 372 // to lookup the service if it's not running, as it will block for a second 373 if (mLogMemoryDealer != 0) { 374 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 375 if (binder != 0) { 376 fdprintf(fd, "\nmedia.log:\n"); 377 Vector<String16> args; 378 binder->dump(fd, args); 379 } 380 } 381 } 382 return NO_ERROR; 383} 384 385sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 386{ 387 // If pid is already in the mClients wp<> map, then use that entry 388 // (for which promote() is always != 0), otherwise create a new entry and Client. 389 sp<Client> client = mClients.valueFor(pid).promote(); 390 if (client == 0) { 391 client = new Client(this, pid); 392 mClients.add(pid, client); 393 } 394 395 return client; 396} 397 398sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 399{ 400 if (mLogMemoryDealer == 0) { 401 return new NBLog::Writer(); 402 } 403 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 404 sp<NBLog::Writer> writer = new NBLog::Writer(size, shared); 405 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 406 if (binder != 0) { 407 interface_cast<IMediaLogService>(binder)->registerWriter(shared, size, name); 408 } 409 return writer; 410} 411 412void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 413{ 414 if (writer == 0) { 415 return; 416 } 417 sp<IMemory> iMemory(writer->getIMemory()); 418 if (iMemory == 0) { 419 return; 420 } 421 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 422 if (binder != 0) { 423 interface_cast<IMediaLogService>(binder)->unregisterWriter(iMemory); 424 // Now the media.log remote reference to IMemory is gone. 425 // When our last local reference to IMemory also drops to zero, 426 // the IMemory destructor will deallocate the region from mMemoryDealer. 427 } 428} 429 430// IAudioFlinger interface 431 432 433sp<IAudioTrack> AudioFlinger::createTrack( 434 audio_stream_type_t streamType, 435 uint32_t sampleRate, 436 audio_format_t format, 437 audio_channel_mask_t channelMask, 438 size_t frameCount, 439 IAudioFlinger::track_flags_t *flags, 440 const sp<IMemory>& sharedBuffer, 441 audio_io_handle_t output, 442 pid_t tid, 443 int *sessionId, 444 String8& name, 445 status_t *status) 446{ 447 sp<PlaybackThread::Track> track; 448 sp<TrackHandle> trackHandle; 449 sp<Client> client; 450 status_t lStatus; 451 int lSessionId; 452 453 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 454 // but if someone uses binder directly they could bypass that and cause us to crash 455 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 456 ALOGE("createTrack() invalid stream type %d", streamType); 457 lStatus = BAD_VALUE; 458 goto Exit; 459 } 460 461 // client is responsible for conversion of 8-bit PCM to 16-bit PCM, 462 // and we don't yet support 8.24 or 32-bit PCM 463 if (audio_is_linear_pcm(format) && format != AUDIO_FORMAT_PCM_16_BIT) { 464 ALOGE("createTrack() invalid format %d", format); 465 lStatus = BAD_VALUE; 466 goto Exit; 467 } 468 469 { 470 Mutex::Autolock _l(mLock); 471 PlaybackThread *thread = checkPlaybackThread_l(output); 472 PlaybackThread *effectThread = NULL; 473 if (thread == NULL) { 474 ALOGE("no playback thread found for output handle %d", output); 475 lStatus = BAD_VALUE; 476 goto Exit; 477 } 478 479 pid_t pid = IPCThreadState::self()->getCallingPid(); 480 client = registerPid_l(pid); 481 482 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 483 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 484 // check if an effect chain with the same session ID is present on another 485 // output thread and move it here. 486 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 487 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 488 if (mPlaybackThreads.keyAt(i) != output) { 489 uint32_t sessions = t->hasAudioSession(*sessionId); 490 if (sessions & PlaybackThread::EFFECT_SESSION) { 491 effectThread = t.get(); 492 break; 493 } 494 } 495 } 496 lSessionId = *sessionId; 497 } else { 498 // if no audio session id is provided, create one here 499 lSessionId = nextUniqueId(); 500 if (sessionId != NULL) { 501 *sessionId = lSessionId; 502 } 503 } 504 ALOGV("createTrack() lSessionId: %d", lSessionId); 505 506 track = thread->createTrack_l(client, streamType, sampleRate, format, 507 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 508 509 // move effect chain to this output thread if an effect on same session was waiting 510 // for a track to be created 511 if (lStatus == NO_ERROR && effectThread != NULL) { 512 Mutex::Autolock _dl(thread->mLock); 513 Mutex::Autolock _sl(effectThread->mLock); 514 moveEffectChain_l(lSessionId, effectThread, thread, true); 515 } 516 517 // Look for sync events awaiting for a session to be used. 518 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 519 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 520 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 521 if (lStatus == NO_ERROR) { 522 (void) track->setSyncEvent(mPendingSyncEvents[i]); 523 } else { 524 mPendingSyncEvents[i]->cancel(); 525 } 526 mPendingSyncEvents.removeAt(i); 527 i--; 528 } 529 } 530 } 531 } 532 if (lStatus == NO_ERROR) { 533 // s for server's pid, n for normal mixer name, f for fast index 534 name = String8::format("s:%d;n:%d;f:%d", getpid_cached, track->name() - AudioMixer::TRACK0, 535 track->fastIndex()); 536 trackHandle = new TrackHandle(track); 537 } else { 538 // remove local strong reference to Client before deleting the Track so that the Client 539 // destructor is called by the TrackBase destructor with mLock held 540 client.clear(); 541 track.clear(); 542 } 543 544Exit: 545 if (status != NULL) { 546 *status = lStatus; 547 } 548 return trackHandle; 549} 550 551uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 552{ 553 Mutex::Autolock _l(mLock); 554 PlaybackThread *thread = checkPlaybackThread_l(output); 555 if (thread == NULL) { 556 ALOGW("sampleRate() unknown thread %d", output); 557 return 0; 558 } 559 return thread->sampleRate(); 560} 561 562int AudioFlinger::channelCount(audio_io_handle_t output) const 563{ 564 Mutex::Autolock _l(mLock); 565 PlaybackThread *thread = checkPlaybackThread_l(output); 566 if (thread == NULL) { 567 ALOGW("channelCount() unknown thread %d", output); 568 return 0; 569 } 570 return thread->channelCount(); 571} 572 573audio_format_t AudioFlinger::format(audio_io_handle_t output) const 574{ 575 Mutex::Autolock _l(mLock); 576 PlaybackThread *thread = checkPlaybackThread_l(output); 577 if (thread == NULL) { 578 ALOGW("format() unknown thread %d", output); 579 return AUDIO_FORMAT_INVALID; 580 } 581 return thread->format(); 582} 583 584size_t AudioFlinger::frameCount(audio_io_handle_t output) const 585{ 586 Mutex::Autolock _l(mLock); 587 PlaybackThread *thread = checkPlaybackThread_l(output); 588 if (thread == NULL) { 589 ALOGW("frameCount() unknown thread %d", output); 590 return 0; 591 } 592 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 593 // should examine all callers and fix them to handle smaller counts 594 return thread->frameCount(); 595} 596 597uint32_t AudioFlinger::latency(audio_io_handle_t output) const 598{ 599 Mutex::Autolock _l(mLock); 600 PlaybackThread *thread = checkPlaybackThread_l(output); 601 if (thread == NULL) { 602 ALOGW("latency(): no playback thread found for output handle %d", output); 603 return 0; 604 } 605 return thread->latency(); 606} 607 608status_t AudioFlinger::setMasterVolume(float value) 609{ 610 status_t ret = initCheck(); 611 if (ret != NO_ERROR) { 612 return ret; 613 } 614 615 // check calling permissions 616 if (!settingsAllowed()) { 617 return PERMISSION_DENIED; 618 } 619 620 Mutex::Autolock _l(mLock); 621 mMasterVolume = value; 622 623 // Set master volume in the HALs which support it. 624 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 625 AutoMutex lock(mHardwareLock); 626 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 627 628 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 629 if (dev->canSetMasterVolume()) { 630 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 631 } 632 mHardwareStatus = AUDIO_HW_IDLE; 633 } 634 635 // Now set the master volume in each playback thread. Playback threads 636 // assigned to HALs which do not have master volume support will apply 637 // master volume during the mix operation. Threads with HALs which do 638 // support master volume will simply ignore the setting. 639 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 640 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 641 642 return NO_ERROR; 643} 644 645status_t AudioFlinger::setMode(audio_mode_t mode) 646{ 647 status_t ret = initCheck(); 648 if (ret != NO_ERROR) { 649 return ret; 650 } 651 652 // check calling permissions 653 if (!settingsAllowed()) { 654 return PERMISSION_DENIED; 655 } 656 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 657 ALOGW("Illegal value: setMode(%d)", mode); 658 return BAD_VALUE; 659 } 660 661 { // scope for the lock 662 AutoMutex lock(mHardwareLock); 663 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 664 mHardwareStatus = AUDIO_HW_SET_MODE; 665 ret = dev->set_mode(dev, mode); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 } 668 669 if (NO_ERROR == ret) { 670 Mutex::Autolock _l(mLock); 671 mMode = mode; 672 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 673 mPlaybackThreads.valueAt(i)->setMode(mode); 674 } 675 676 return ret; 677} 678 679status_t AudioFlinger::setMicMute(bool state) 680{ 681 status_t ret = initCheck(); 682 if (ret != NO_ERROR) { 683 return ret; 684 } 685 686 // check calling permissions 687 if (!settingsAllowed()) { 688 return PERMISSION_DENIED; 689 } 690 691 AutoMutex lock(mHardwareLock); 692 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 693 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 694 ret = dev->set_mic_mute(dev, state); 695 mHardwareStatus = AUDIO_HW_IDLE; 696 return ret; 697} 698 699bool AudioFlinger::getMicMute() const 700{ 701 status_t ret = initCheck(); 702 if (ret != NO_ERROR) { 703 return false; 704 } 705 706 bool state = AUDIO_MODE_INVALID; 707 AutoMutex lock(mHardwareLock); 708 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 709 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 710 dev->get_mic_mute(dev, &state); 711 mHardwareStatus = AUDIO_HW_IDLE; 712 return state; 713} 714 715status_t AudioFlinger::setMasterMute(bool muted) 716{ 717 status_t ret = initCheck(); 718 if (ret != NO_ERROR) { 719 return ret; 720 } 721 722 // check calling permissions 723 if (!settingsAllowed()) { 724 return PERMISSION_DENIED; 725 } 726 727 Mutex::Autolock _l(mLock); 728 mMasterMute = muted; 729 730 // Set master mute in the HALs which support it. 731 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 732 AutoMutex lock(mHardwareLock); 733 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 734 735 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 736 if (dev->canSetMasterMute()) { 737 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 738 } 739 mHardwareStatus = AUDIO_HW_IDLE; 740 } 741 742 // Now set the master mute in each playback thread. Playback threads 743 // assigned to HALs which do not have master mute support will apply master 744 // mute during the mix operation. Threads with HALs which do support master 745 // mute will simply ignore the setting. 746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 747 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 748 749 return NO_ERROR; 750} 751 752float AudioFlinger::masterVolume() const 753{ 754 Mutex::Autolock _l(mLock); 755 return masterVolume_l(); 756} 757 758bool AudioFlinger::masterMute() const 759{ 760 Mutex::Autolock _l(mLock); 761 return masterMute_l(); 762} 763 764float AudioFlinger::masterVolume_l() const 765{ 766 return mMasterVolume; 767} 768 769bool AudioFlinger::masterMute_l() const 770{ 771 return mMasterMute; 772} 773 774status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 775 audio_io_handle_t output) 776{ 777 // check calling permissions 778 if (!settingsAllowed()) { 779 return PERMISSION_DENIED; 780 } 781 782 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 783 ALOGE("setStreamVolume() invalid stream %d", stream); 784 return BAD_VALUE; 785 } 786 787 AutoMutex lock(mLock); 788 PlaybackThread *thread = NULL; 789 if (output) { 790 thread = checkPlaybackThread_l(output); 791 if (thread == NULL) { 792 return BAD_VALUE; 793 } 794 } 795 796 mStreamTypes[stream].volume = value; 797 798 if (thread == NULL) { 799 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 800 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 801 } 802 } else { 803 thread->setStreamVolume(stream, value); 804 } 805 806 return NO_ERROR; 807} 808 809status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 810{ 811 // check calling permissions 812 if (!settingsAllowed()) { 813 return PERMISSION_DENIED; 814 } 815 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 817 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 818 ALOGE("setStreamMute() invalid stream %d", stream); 819 return BAD_VALUE; 820 } 821 822 AutoMutex lock(mLock); 823 mStreamTypes[stream].mute = muted; 824 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 825 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 826 827 return NO_ERROR; 828} 829 830float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return 0.0f; 834 } 835 836 AutoMutex lock(mLock); 837 float volume; 838 if (output) { 839 PlaybackThread *thread = checkPlaybackThread_l(output); 840 if (thread == NULL) { 841 return 0.0f; 842 } 843 volume = thread->streamVolume(stream); 844 } else { 845 volume = streamVolume_l(stream); 846 } 847 848 return volume; 849} 850 851bool AudioFlinger::streamMute(audio_stream_type_t stream) const 852{ 853 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 854 return true; 855 } 856 857 AutoMutex lock(mLock); 858 return streamMute_l(stream); 859} 860 861status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 862{ 863 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 864 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 865 866 // check calling permissions 867 if (!settingsAllowed()) { 868 return PERMISSION_DENIED; 869 } 870 871 // ioHandle == 0 means the parameters are global to the audio hardware interface 872 if (ioHandle == 0) { 873 Mutex::Autolock _l(mLock); 874 status_t final_result = NO_ERROR; 875 { 876 AutoMutex lock(mHardwareLock); 877 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 878 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 879 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 880 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 881 final_result = result ?: final_result; 882 } 883 mHardwareStatus = AUDIO_HW_IDLE; 884 } 885 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 886 AudioParameter param = AudioParameter(keyValuePairs); 887 String8 value; 888 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 889 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 890 if (mBtNrecIsOff != btNrecIsOff) { 891 for (size_t i = 0; i < mRecordThreads.size(); i++) { 892 sp<RecordThread> thread = mRecordThreads.valueAt(i); 893 audio_devices_t device = thread->inDevice(); 894 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 895 // collect all of the thread's session IDs 896 KeyedVector<int, bool> ids = thread->sessionIds(); 897 // suspend effects associated with those session IDs 898 for (size_t j = 0; j < ids.size(); ++j) { 899 int sessionId = ids.keyAt(j); 900 thread->setEffectSuspended(FX_IID_AEC, 901 suspend, 902 sessionId); 903 thread->setEffectSuspended(FX_IID_NS, 904 suspend, 905 sessionId); 906 } 907 } 908 mBtNrecIsOff = btNrecIsOff; 909 } 910 } 911 String8 screenState; 912 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 913 bool isOff = screenState == "off"; 914 if (isOff != (AudioFlinger::mScreenState & 1)) { 915 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 916 } 917 } 918 return final_result; 919 } 920 921 // hold a strong ref on thread in case closeOutput() or closeInput() is called 922 // and the thread is exited once the lock is released 923 sp<ThreadBase> thread; 924 { 925 Mutex::Autolock _l(mLock); 926 thread = checkPlaybackThread_l(ioHandle); 927 if (thread == 0) { 928 thread = checkRecordThread_l(ioHandle); 929 } else if (thread == primaryPlaybackThread_l()) { 930 // indicate output device change to all input threads for pre processing 931 AudioParameter param = AudioParameter(keyValuePairs); 932 int value; 933 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 934 (value != 0)) { 935 for (size_t i = 0; i < mRecordThreads.size(); i++) { 936 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 937 } 938 } 939 } 940 } 941 if (thread != 0) { 942 return thread->setParameters(keyValuePairs); 943 } 944 return BAD_VALUE; 945} 946 947String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 948{ 949 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 950 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 951 952 Mutex::Autolock _l(mLock); 953 954 if (ioHandle == 0) { 955 String8 out_s8; 956 957 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 958 char *s; 959 { 960 AutoMutex lock(mHardwareLock); 961 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 962 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 963 s = dev->get_parameters(dev, keys.string()); 964 mHardwareStatus = AUDIO_HW_IDLE; 965 } 966 out_s8 += String8(s ? s : ""); 967 free(s); 968 } 969 return out_s8; 970 } 971 972 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 973 if (playbackThread != NULL) { 974 return playbackThread->getParameters(keys); 975 } 976 RecordThread *recordThread = checkRecordThread_l(ioHandle); 977 if (recordThread != NULL) { 978 return recordThread->getParameters(keys); 979 } 980 return String8(""); 981} 982 983size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 984 audio_channel_mask_t channelMask) const 985{ 986 status_t ret = initCheck(); 987 if (ret != NO_ERROR) { 988 return 0; 989 } 990 991 AutoMutex lock(mHardwareLock); 992 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 993 struct audio_config config; 994 memset(&config, 0, sizeof(config)); 995 config.sample_rate = sampleRate; 996 config.channel_mask = channelMask; 997 config.format = format; 998 999 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1000 size_t size = dev->get_input_buffer_size(dev, &config); 1001 mHardwareStatus = AUDIO_HW_IDLE; 1002 return size; 1003} 1004 1005unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1006{ 1007 Mutex::Autolock _l(mLock); 1008 1009 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1010 if (recordThread != NULL) { 1011 return recordThread->getInputFramesLost(); 1012 } 1013 return 0; 1014} 1015 1016status_t AudioFlinger::setVoiceVolume(float value) 1017{ 1018 status_t ret = initCheck(); 1019 if (ret != NO_ERROR) { 1020 return ret; 1021 } 1022 1023 // check calling permissions 1024 if (!settingsAllowed()) { 1025 return PERMISSION_DENIED; 1026 } 1027 1028 AutoMutex lock(mHardwareLock); 1029 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1030 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1031 ret = dev->set_voice_volume(dev, value); 1032 mHardwareStatus = AUDIO_HW_IDLE; 1033 1034 return ret; 1035} 1036 1037status_t AudioFlinger::getRenderPosition(size_t *halFrames, size_t *dspFrames, 1038 audio_io_handle_t output) const 1039{ 1040 status_t status; 1041 1042 Mutex::Autolock _l(mLock); 1043 1044 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1045 if (playbackThread != NULL) { 1046 return playbackThread->getRenderPosition(halFrames, dspFrames); 1047 } 1048 1049 return BAD_VALUE; 1050} 1051 1052void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1053{ 1054 1055 Mutex::Autolock _l(mLock); 1056 1057 pid_t pid = IPCThreadState::self()->getCallingPid(); 1058 if (mNotificationClients.indexOfKey(pid) < 0) { 1059 sp<NotificationClient> notificationClient = new NotificationClient(this, 1060 client, 1061 pid); 1062 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1063 1064 mNotificationClients.add(pid, notificationClient); 1065 1066 sp<IBinder> binder = client->asBinder(); 1067 binder->linkToDeath(notificationClient); 1068 1069 // the config change is always sent from playback or record threads to avoid deadlock 1070 // with AudioSystem::gLock 1071 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1072 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1073 } 1074 1075 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1076 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1077 } 1078 } 1079} 1080 1081void AudioFlinger::removeNotificationClient(pid_t pid) 1082{ 1083 Mutex::Autolock _l(mLock); 1084 1085 mNotificationClients.removeItem(pid); 1086 1087 ALOGV("%d died, releasing its sessions", pid); 1088 size_t num = mAudioSessionRefs.size(); 1089 bool removed = false; 1090 for (size_t i = 0; i< num; ) { 1091 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1092 ALOGV(" pid %d @ %d", ref->mPid, i); 1093 if (ref->mPid == pid) { 1094 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1095 mAudioSessionRefs.removeAt(i); 1096 delete ref; 1097 removed = true; 1098 num--; 1099 } else { 1100 i++; 1101 } 1102 } 1103 if (removed) { 1104 purgeStaleEffects_l(); 1105 } 1106} 1107 1108// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1109void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1110{ 1111 size_t size = mNotificationClients.size(); 1112 for (size_t i = 0; i < size; i++) { 1113 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1114 param2); 1115 } 1116} 1117 1118// removeClient_l() must be called with AudioFlinger::mLock held 1119void AudioFlinger::removeClient_l(pid_t pid) 1120{ 1121 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1122 IPCThreadState::self()->getCallingPid()); 1123 mClients.removeItem(pid); 1124} 1125 1126// getEffectThread_l() must be called with AudioFlinger::mLock held 1127sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1128{ 1129 sp<PlaybackThread> thread; 1130 1131 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1132 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1133 ALOG_ASSERT(thread == 0); 1134 thread = mPlaybackThreads.valueAt(i); 1135 } 1136 } 1137 1138 return thread; 1139} 1140 1141 1142 1143// ---------------------------------------------------------------------------- 1144 1145AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1146 : RefBase(), 1147 mAudioFlinger(audioFlinger), 1148 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1149 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1150 mPid(pid), 1151 mTimedTrackCount(0) 1152{ 1153 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1154} 1155 1156// Client destructor must be called with AudioFlinger::mLock held 1157AudioFlinger::Client::~Client() 1158{ 1159 mAudioFlinger->removeClient_l(mPid); 1160} 1161 1162sp<MemoryDealer> AudioFlinger::Client::heap() const 1163{ 1164 return mMemoryDealer; 1165} 1166 1167// Reserve one of the limited slots for a timed audio track associated 1168// with this client 1169bool AudioFlinger::Client::reserveTimedTrack() 1170{ 1171 const int kMaxTimedTracksPerClient = 4; 1172 1173 Mutex::Autolock _l(mTimedTrackLock); 1174 1175 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1176 ALOGW("can not create timed track - pid %d has exceeded the limit", 1177 mPid); 1178 return false; 1179 } 1180 1181 mTimedTrackCount++; 1182 return true; 1183} 1184 1185// Release a slot for a timed audio track 1186void AudioFlinger::Client::releaseTimedTrack() 1187{ 1188 Mutex::Autolock _l(mTimedTrackLock); 1189 mTimedTrackCount--; 1190} 1191 1192// ---------------------------------------------------------------------------- 1193 1194AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1195 const sp<IAudioFlingerClient>& client, 1196 pid_t pid) 1197 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1198{ 1199} 1200 1201AudioFlinger::NotificationClient::~NotificationClient() 1202{ 1203} 1204 1205void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 1206{ 1207 sp<NotificationClient> keep(this); 1208 mAudioFlinger->removeNotificationClient(mPid); 1209} 1210 1211 1212// ---------------------------------------------------------------------------- 1213 1214static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1215 return audio_is_remote_submix_device(inDevice); 1216} 1217 1218sp<IAudioRecord> AudioFlinger::openRecord( 1219 audio_io_handle_t input, 1220 uint32_t sampleRate, 1221 audio_format_t format, 1222 audio_channel_mask_t channelMask, 1223 size_t frameCount, 1224 IAudioFlinger::track_flags_t *flags, 1225 pid_t tid, 1226 int *sessionId, 1227 status_t *status) 1228{ 1229 sp<RecordThread::RecordTrack> recordTrack; 1230 sp<RecordHandle> recordHandle; 1231 sp<Client> client; 1232 status_t lStatus; 1233 RecordThread *thread; 1234 size_t inFrameCount; 1235 int lSessionId; 1236 1237 // check calling permissions 1238 if (!recordingAllowed()) { 1239 lStatus = PERMISSION_DENIED; 1240 goto Exit; 1241 } 1242 1243 if (format != AUDIO_FORMAT_PCM_16_BIT) { 1244 ALOGE("openRecord() invalid format %d", format); 1245 lStatus = BAD_VALUE; 1246 goto Exit; 1247 } 1248 1249 // add client to list 1250 { // scope for mLock 1251 Mutex::Autolock _l(mLock); 1252 thread = checkRecordThread_l(input); 1253 if (thread == NULL) { 1254 lStatus = BAD_VALUE; 1255 goto Exit; 1256 } 1257 1258 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1259 && !captureAudioOutputAllowed()) { 1260 lStatus = PERMISSION_DENIED; 1261 goto Exit; 1262 } 1263 1264 pid_t pid = IPCThreadState::self()->getCallingPid(); 1265 client = registerPid_l(pid); 1266 1267 // If no audio session id is provided, create one here 1268 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1269 lSessionId = *sessionId; 1270 } else { 1271 lSessionId = nextUniqueId(); 1272 if (sessionId != NULL) { 1273 *sessionId = lSessionId; 1274 } 1275 } 1276 // create new record track. 1277 // The record track uses one track in mHardwareMixerThread by convention. 1278 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1279 frameCount, lSessionId, flags, tid, &lStatus); 1280 } 1281 if (lStatus != NO_ERROR) { 1282 // remove local strong reference to Client before deleting the RecordTrack so that the 1283 // Client destructor is called by the TrackBase destructor with mLock held 1284 client.clear(); 1285 recordTrack.clear(); 1286 goto Exit; 1287 } 1288 1289 // return to handle to client 1290 recordHandle = new RecordHandle(recordTrack); 1291 lStatus = NO_ERROR; 1292 1293Exit: 1294 if (status) { 1295 *status = lStatus; 1296 } 1297 return recordHandle; 1298} 1299 1300 1301 1302// ---------------------------------------------------------------------------- 1303 1304audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1305{ 1306 if (!settingsAllowed()) { 1307 return 0; 1308 } 1309 Mutex::Autolock _l(mLock); 1310 return loadHwModule_l(name); 1311} 1312 1313// loadHwModule_l() must be called with AudioFlinger::mLock held 1314audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1315{ 1316 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1317 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1318 ALOGW("loadHwModule() module %s already loaded", name); 1319 return mAudioHwDevs.keyAt(i); 1320 } 1321 } 1322 1323 audio_hw_device_t *dev; 1324 1325 int rc = load_audio_interface(name, &dev); 1326 if (rc) { 1327 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1328 return 0; 1329 } 1330 1331 mHardwareStatus = AUDIO_HW_INIT; 1332 rc = dev->init_check(dev); 1333 mHardwareStatus = AUDIO_HW_IDLE; 1334 if (rc) { 1335 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1336 return 0; 1337 } 1338 1339 // Check and cache this HAL's level of support for master mute and master 1340 // volume. If this is the first HAL opened, and it supports the get 1341 // methods, use the initial values provided by the HAL as the current 1342 // master mute and volume settings. 1343 1344 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1345 { // scope for auto-lock pattern 1346 AutoMutex lock(mHardwareLock); 1347 1348 if (0 == mAudioHwDevs.size()) { 1349 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1350 if (NULL != dev->get_master_volume) { 1351 float mv; 1352 if (OK == dev->get_master_volume(dev, &mv)) { 1353 mMasterVolume = mv; 1354 } 1355 } 1356 1357 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1358 if (NULL != dev->get_master_mute) { 1359 bool mm; 1360 if (OK == dev->get_master_mute(dev, &mm)) { 1361 mMasterMute = mm; 1362 } 1363 } 1364 } 1365 1366 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1367 if ((NULL != dev->set_master_volume) && 1368 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1369 flags = static_cast<AudioHwDevice::Flags>(flags | 1370 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1371 } 1372 1373 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1374 if ((NULL != dev->set_master_mute) && 1375 (OK == dev->set_master_mute(dev, mMasterMute))) { 1376 flags = static_cast<AudioHwDevice::Flags>(flags | 1377 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1378 } 1379 1380 mHardwareStatus = AUDIO_HW_IDLE; 1381 } 1382 1383 audio_module_handle_t handle = nextUniqueId(); 1384 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1385 1386 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1387 name, dev->common.module->name, dev->common.module->id, handle); 1388 1389 return handle; 1390 1391} 1392 1393// ---------------------------------------------------------------------------- 1394 1395uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1396{ 1397 Mutex::Autolock _l(mLock); 1398 PlaybackThread *thread = primaryPlaybackThread_l(); 1399 return thread != NULL ? thread->sampleRate() : 0; 1400} 1401 1402size_t AudioFlinger::getPrimaryOutputFrameCount() 1403{ 1404 Mutex::Autolock _l(mLock); 1405 PlaybackThread *thread = primaryPlaybackThread_l(); 1406 return thread != NULL ? thread->frameCountHAL() : 0; 1407} 1408 1409// ---------------------------------------------------------------------------- 1410 1411status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1412{ 1413 uid_t uid = IPCThreadState::self()->getCallingUid(); 1414 if (uid != AID_SYSTEM) { 1415 return PERMISSION_DENIED; 1416 } 1417 Mutex::Autolock _l(mLock); 1418 if (mIsDeviceTypeKnown) { 1419 return INVALID_OPERATION; 1420 } 1421 mIsLowRamDevice = isLowRamDevice; 1422 mIsDeviceTypeKnown = true; 1423 return NO_ERROR; 1424} 1425 1426// ---------------------------------------------------------------------------- 1427 1428audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1429 audio_devices_t *pDevices, 1430 uint32_t *pSamplingRate, 1431 audio_format_t *pFormat, 1432 audio_channel_mask_t *pChannelMask, 1433 uint32_t *pLatencyMs, 1434 audio_output_flags_t flags, 1435 const audio_offload_info_t *offloadInfo) 1436{ 1437 PlaybackThread *thread = NULL; 1438 struct audio_config config; 1439 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1440 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1441 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1442 if (offloadInfo) { 1443 config.offload_info = *offloadInfo; 1444 } 1445 1446 audio_stream_out_t *outStream = NULL; 1447 AudioHwDevice *outHwDev; 1448 1449 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1450 module, 1451 (pDevices != NULL) ? *pDevices : 0, 1452 config.sample_rate, 1453 config.format, 1454 config.channel_mask, 1455 flags); 1456 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1457 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version ); 1458 1459 if (pDevices == NULL || *pDevices == 0) { 1460 return 0; 1461 } 1462 1463 Mutex::Autolock _l(mLock); 1464 1465 outHwDev = findSuitableHwDev_l(module, *pDevices); 1466 if (outHwDev == NULL) 1467 return 0; 1468 1469 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1470 audio_io_handle_t id = nextUniqueId(); 1471 1472 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1473 1474 status_t status = hwDevHal->open_output_stream(hwDevHal, 1475 id, 1476 *pDevices, 1477 (audio_output_flags_t)flags, 1478 &config, 1479 &outStream); 1480 1481 mHardwareStatus = AUDIO_HW_IDLE; 1482 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1483 "Channels %x, status %d", 1484 outStream, 1485 config.sample_rate, 1486 config.format, 1487 config.channel_mask, 1488 status); 1489 1490 if (status == NO_ERROR && outStream != NULL) { 1491 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1492 1493 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1494 thread = new OffloadThread(this, output, id, *pDevices); 1495 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1496 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 1497 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 1498 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1499 thread = new DirectOutputThread(this, output, id, *pDevices); 1500 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1501 } else { 1502 thread = new MixerThread(this, output, id, *pDevices); 1503 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1504 } 1505 mPlaybackThreads.add(id, thread); 1506 1507 if (pSamplingRate != NULL) { 1508 *pSamplingRate = config.sample_rate; 1509 } 1510 if (pFormat != NULL) { 1511 *pFormat = config.format; 1512 } 1513 if (pChannelMask != NULL) { 1514 *pChannelMask = config.channel_mask; 1515 } 1516 if (pLatencyMs != NULL) { 1517 *pLatencyMs = thread->latency(); 1518 } 1519 1520 // notify client processes of the new output creation 1521 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1522 1523 // the first primary output opened designates the primary hw device 1524 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1525 ALOGI("Using module %d has the primary audio interface", module); 1526 mPrimaryHardwareDev = outHwDev; 1527 1528 AutoMutex lock(mHardwareLock); 1529 mHardwareStatus = AUDIO_HW_SET_MODE; 1530 hwDevHal->set_mode(hwDevHal, mMode); 1531 mHardwareStatus = AUDIO_HW_IDLE; 1532 } 1533 return id; 1534 } 1535 1536 return 0; 1537} 1538 1539audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1540 audio_io_handle_t output2) 1541{ 1542 Mutex::Autolock _l(mLock); 1543 MixerThread *thread1 = checkMixerThread_l(output1); 1544 MixerThread *thread2 = checkMixerThread_l(output2); 1545 1546 if (thread1 == NULL || thread2 == NULL) { 1547 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1548 output2); 1549 return 0; 1550 } 1551 1552 audio_io_handle_t id = nextUniqueId(); 1553 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1554 thread->addOutputTrack(thread2); 1555 mPlaybackThreads.add(id, thread); 1556 // notify client processes of the new output creation 1557 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 1558 return id; 1559} 1560 1561status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1562{ 1563 return closeOutput_nonvirtual(output); 1564} 1565 1566status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1567{ 1568 // keep strong reference on the playback thread so that 1569 // it is not destroyed while exit() is executed 1570 sp<PlaybackThread> thread; 1571 { 1572 Mutex::Autolock _l(mLock); 1573 thread = checkPlaybackThread_l(output); 1574 if (thread == NULL) { 1575 return BAD_VALUE; 1576 } 1577 1578 ALOGV("closeOutput() %d", output); 1579 1580 if (thread->type() == ThreadBase::MIXER) { 1581 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1582 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1583 DuplicatingThread *dupThread = 1584 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1585 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1586 1587 } 1588 } 1589 } 1590 1591 1592 mPlaybackThreads.removeItem(output); 1593 // save all effects to the default thread 1594 if (mPlaybackThreads.size()) { 1595 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1596 if (dstThread != NULL) { 1597 // audioflinger lock is held here so the acquisition order of thread locks does not 1598 // matter 1599 Mutex::Autolock _dl(dstThread->mLock); 1600 Mutex::Autolock _sl(thread->mLock); 1601 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1602 for (size_t i = 0; i < effectChains.size(); i ++) { 1603 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1604 } 1605 } 1606 } 1607 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 1608 } 1609 thread->exit(); 1610 // The thread entity (active unit of execution) is no longer running here, 1611 // but the ThreadBase container still exists. 1612 1613 if (thread->type() != ThreadBase::DUPLICATING) { 1614 AudioStreamOut *out = thread->clearOutput(); 1615 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1616 // from now on thread->mOutput is NULL 1617 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1618 delete out; 1619 } 1620 return NO_ERROR; 1621} 1622 1623status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1624{ 1625 Mutex::Autolock _l(mLock); 1626 PlaybackThread *thread = checkPlaybackThread_l(output); 1627 1628 if (thread == NULL) { 1629 return BAD_VALUE; 1630 } 1631 1632 ALOGV("suspendOutput() %d", output); 1633 thread->suspend(); 1634 1635 return NO_ERROR; 1636} 1637 1638status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1639{ 1640 Mutex::Autolock _l(mLock); 1641 PlaybackThread *thread = checkPlaybackThread_l(output); 1642 1643 if (thread == NULL) { 1644 return BAD_VALUE; 1645 } 1646 1647 ALOGV("restoreOutput() %d", output); 1648 1649 thread->restore(); 1650 1651 return NO_ERROR; 1652} 1653 1654audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1655 audio_devices_t *pDevices, 1656 uint32_t *pSamplingRate, 1657 audio_format_t *pFormat, 1658 audio_channel_mask_t *pChannelMask) 1659{ 1660 status_t status; 1661 RecordThread *thread = NULL; 1662 struct audio_config config; 1663 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1664 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1665 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1666 1667 uint32_t reqSamplingRate = config.sample_rate; 1668 audio_format_t reqFormat = config.format; 1669 audio_channel_mask_t reqChannels = config.channel_mask; 1670 audio_stream_in_t *inStream = NULL; 1671 AudioHwDevice *inHwDev; 1672 1673 if (pDevices == NULL || *pDevices == 0) { 1674 return 0; 1675 } 1676 1677 Mutex::Autolock _l(mLock); 1678 1679 inHwDev = findSuitableHwDev_l(module, *pDevices); 1680 if (inHwDev == NULL) 1681 return 0; 1682 1683 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1684 audio_io_handle_t id = nextUniqueId(); 1685 1686 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1687 &inStream); 1688 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, " 1689 "status %d", 1690 inStream, 1691 config.sample_rate, 1692 config.format, 1693 config.channel_mask, 1694 status); 1695 1696 // If the input could not be opened with the requested parameters and we can handle the 1697 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1698 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1699 if (status == BAD_VALUE && 1700 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1701 (config.sample_rate <= 2 * reqSamplingRate) && 1702 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 1703 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1704 inStream = NULL; 1705 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1706 } 1707 1708 if (status == NO_ERROR && inStream != NULL) { 1709 1710#ifdef TEE_SINK 1711 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1712 // or (re-)create if current Pipe is idle and does not match the new format 1713 sp<NBAIO_Sink> teeSink; 1714 enum { 1715 TEE_SINK_NO, // don't copy input 1716 TEE_SINK_NEW, // copy input using a new pipe 1717 TEE_SINK_OLD, // copy input using an existing pipe 1718 } kind; 1719 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1720 popcount(inStream->common.get_channels(&inStream->common))); 1721 if (!mTeeSinkInputEnabled) { 1722 kind = TEE_SINK_NO; 1723 } else if (format == Format_Invalid) { 1724 kind = TEE_SINK_NO; 1725 } else if (mRecordTeeSink == 0) { 1726 kind = TEE_SINK_NEW; 1727 } else if (mRecordTeeSink->getStrongCount() != 1) { 1728 kind = TEE_SINK_NO; 1729 } else if (format == mRecordTeeSink->format()) { 1730 kind = TEE_SINK_OLD; 1731 } else { 1732 kind = TEE_SINK_NEW; 1733 } 1734 switch (kind) { 1735 case TEE_SINK_NEW: { 1736 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1737 size_t numCounterOffers = 0; 1738 const NBAIO_Format offers[1] = {format}; 1739 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1740 ALOG_ASSERT(index == 0); 1741 PipeReader *pipeReader = new PipeReader(*pipe); 1742 numCounterOffers = 0; 1743 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1744 ALOG_ASSERT(index == 0); 1745 mRecordTeeSink = pipe; 1746 mRecordTeeSource = pipeReader; 1747 teeSink = pipe; 1748 } 1749 break; 1750 case TEE_SINK_OLD: 1751 teeSink = mRecordTeeSink; 1752 break; 1753 case TEE_SINK_NO: 1754 default: 1755 break; 1756 } 1757#endif 1758 1759 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1760 1761 // Start record thread 1762 // RecordThread requires both input and output device indication to forward to audio 1763 // pre processing modules 1764 thread = new RecordThread(this, 1765 input, 1766 reqSamplingRate, 1767 reqChannels, 1768 id, 1769 primaryOutputDevice_l(), 1770 *pDevices 1771#ifdef TEE_SINK 1772 , teeSink 1773#endif 1774 ); 1775 mRecordThreads.add(id, thread); 1776 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1777 if (pSamplingRate != NULL) { 1778 *pSamplingRate = reqSamplingRate; 1779 } 1780 if (pFormat != NULL) { 1781 *pFormat = config.format; 1782 } 1783 if (pChannelMask != NULL) { 1784 *pChannelMask = reqChannels; 1785 } 1786 1787 // notify client processes of the new input creation 1788 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 1789 return id; 1790 } 1791 1792 return 0; 1793} 1794 1795status_t AudioFlinger::closeInput(audio_io_handle_t input) 1796{ 1797 return closeInput_nonvirtual(input); 1798} 1799 1800status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1801{ 1802 // keep strong reference on the record thread so that 1803 // it is not destroyed while exit() is executed 1804 sp<RecordThread> thread; 1805 { 1806 Mutex::Autolock _l(mLock); 1807 thread = checkRecordThread_l(input); 1808 if (thread == 0) { 1809 return BAD_VALUE; 1810 } 1811 1812 ALOGV("closeInput() %d", input); 1813 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 1814 mRecordThreads.removeItem(input); 1815 } 1816 thread->exit(); 1817 // The thread entity (active unit of execution) is no longer running here, 1818 // but the ThreadBase container still exists. 1819 1820 AudioStreamIn *in = thread->clearInput(); 1821 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1822 // from now on thread->mInput is NULL 1823 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1824 delete in; 1825 1826 return NO_ERROR; 1827} 1828 1829status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 1830{ 1831 Mutex::Autolock _l(mLock); 1832 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 1833 1834 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1835 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1836 thread->invalidateTracks(stream); 1837 } 1838 1839 return NO_ERROR; 1840} 1841 1842 1843int AudioFlinger::newAudioSessionId() 1844{ 1845 return nextUniqueId(); 1846} 1847 1848void AudioFlinger::acquireAudioSessionId(int audioSession) 1849{ 1850 Mutex::Autolock _l(mLock); 1851 pid_t caller = IPCThreadState::self()->getCallingPid(); 1852 ALOGV("acquiring %d from %d", audioSession, caller); 1853 size_t num = mAudioSessionRefs.size(); 1854 for (size_t i = 0; i< num; i++) { 1855 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 1856 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1857 ref->mCnt++; 1858 ALOGV(" incremented refcount to %d", ref->mCnt); 1859 return; 1860 } 1861 } 1862 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 1863 ALOGV(" added new entry for %d", audioSession); 1864} 1865 1866void AudioFlinger::releaseAudioSessionId(int audioSession) 1867{ 1868 Mutex::Autolock _l(mLock); 1869 pid_t caller = IPCThreadState::self()->getCallingPid(); 1870 ALOGV("releasing %d from %d", audioSession, caller); 1871 size_t num = mAudioSessionRefs.size(); 1872 for (size_t i = 0; i< num; i++) { 1873 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1874 if (ref->mSessionid == audioSession && ref->mPid == caller) { 1875 ref->mCnt--; 1876 ALOGV(" decremented refcount to %d", ref->mCnt); 1877 if (ref->mCnt == 0) { 1878 mAudioSessionRefs.removeAt(i); 1879 delete ref; 1880 purgeStaleEffects_l(); 1881 } 1882 return; 1883 } 1884 } 1885 ALOGW("session id %d not found for pid %d", audioSession, caller); 1886} 1887 1888void AudioFlinger::purgeStaleEffects_l() { 1889 1890 ALOGV("purging stale effects"); 1891 1892 Vector< sp<EffectChain> > chains; 1893 1894 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1895 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 1896 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1897 sp<EffectChain> ec = t->mEffectChains[j]; 1898 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 1899 chains.push(ec); 1900 } 1901 } 1902 } 1903 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1904 sp<RecordThread> t = mRecordThreads.valueAt(i); 1905 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 1906 sp<EffectChain> ec = t->mEffectChains[j]; 1907 chains.push(ec); 1908 } 1909 } 1910 1911 for (size_t i = 0; i < chains.size(); i++) { 1912 sp<EffectChain> ec = chains[i]; 1913 int sessionid = ec->sessionId(); 1914 sp<ThreadBase> t = ec->mThread.promote(); 1915 if (t == 0) { 1916 continue; 1917 } 1918 size_t numsessionrefs = mAudioSessionRefs.size(); 1919 bool found = false; 1920 for (size_t k = 0; k < numsessionrefs; k++) { 1921 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 1922 if (ref->mSessionid == sessionid) { 1923 ALOGV(" session %d still exists for %d with %d refs", 1924 sessionid, ref->mPid, ref->mCnt); 1925 found = true; 1926 break; 1927 } 1928 } 1929 if (!found) { 1930 Mutex::Autolock _l (t->mLock); 1931 // remove all effects from the chain 1932 while (ec->mEffects.size()) { 1933 sp<EffectModule> effect = ec->mEffects[0]; 1934 effect->unPin(); 1935 t->removeEffect_l(effect); 1936 if (effect->purgeHandles()) { 1937 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 1938 } 1939 AudioSystem::unregisterEffect(effect->id()); 1940 } 1941 } 1942 } 1943 return; 1944} 1945 1946// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 1947AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 1948{ 1949 return mPlaybackThreads.valueFor(output).get(); 1950} 1951 1952// checkMixerThread_l() must be called with AudioFlinger::mLock held 1953AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 1954{ 1955 PlaybackThread *thread = checkPlaybackThread_l(output); 1956 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 1957} 1958 1959// checkRecordThread_l() must be called with AudioFlinger::mLock held 1960AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 1961{ 1962 return mRecordThreads.valueFor(input).get(); 1963} 1964 1965uint32_t AudioFlinger::nextUniqueId() 1966{ 1967 return android_atomic_inc(&mNextUniqueId); 1968} 1969 1970AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 1971{ 1972 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1973 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 1974 AudioStreamOut *output = thread->getOutput(); 1975 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 1976 return thread; 1977 } 1978 } 1979 return NULL; 1980} 1981 1982audio_devices_t AudioFlinger::primaryOutputDevice_l() const 1983{ 1984 PlaybackThread *thread = primaryPlaybackThread_l(); 1985 1986 if (thread == NULL) { 1987 return 0; 1988 } 1989 1990 return thread->outDevice(); 1991} 1992 1993sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 1994 int triggerSession, 1995 int listenerSession, 1996 sync_event_callback_t callBack, 1997 void *cookie) 1998{ 1999 Mutex::Autolock _l(mLock); 2000 2001 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2002 status_t playStatus = NAME_NOT_FOUND; 2003 status_t recStatus = NAME_NOT_FOUND; 2004 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2005 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2006 if (playStatus == NO_ERROR) { 2007 return event; 2008 } 2009 } 2010 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2011 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2012 if (recStatus == NO_ERROR) { 2013 return event; 2014 } 2015 } 2016 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2017 mPendingSyncEvents.add(event); 2018 } else { 2019 ALOGV("createSyncEvent() invalid event %d", event->type()); 2020 event.clear(); 2021 } 2022 return event; 2023} 2024 2025// ---------------------------------------------------------------------------- 2026// Effect management 2027// ---------------------------------------------------------------------------- 2028 2029 2030status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2031{ 2032 Mutex::Autolock _l(mLock); 2033 return EffectQueryNumberEffects(numEffects); 2034} 2035 2036status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2037{ 2038 Mutex::Autolock _l(mLock); 2039 return EffectQueryEffect(index, descriptor); 2040} 2041 2042status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2043 effect_descriptor_t *descriptor) const 2044{ 2045 Mutex::Autolock _l(mLock); 2046 return EffectGetDescriptor(pUuid, descriptor); 2047} 2048 2049 2050sp<IEffect> AudioFlinger::createEffect( 2051 effect_descriptor_t *pDesc, 2052 const sp<IEffectClient>& effectClient, 2053 int32_t priority, 2054 audio_io_handle_t io, 2055 int sessionId, 2056 status_t *status, 2057 int *id, 2058 int *enabled) 2059{ 2060 status_t lStatus = NO_ERROR; 2061 sp<EffectHandle> handle; 2062 effect_descriptor_t desc; 2063 2064 pid_t pid = IPCThreadState::self()->getCallingPid(); 2065 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2066 pid, effectClient.get(), priority, sessionId, io); 2067 2068 if (pDesc == NULL) { 2069 lStatus = BAD_VALUE; 2070 goto Exit; 2071 } 2072 2073 // check audio settings permission for global effects 2074 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2075 lStatus = PERMISSION_DENIED; 2076 goto Exit; 2077 } 2078 2079 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2080 // that can only be created by audio policy manager (running in same process) 2081 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2082 lStatus = PERMISSION_DENIED; 2083 goto Exit; 2084 } 2085 2086 { 2087 Mutex::Autolock _l(mLock); 2088 2089 2090 if (!EffectIsNullUuid(&pDesc->uuid)) { 2091 // if uuid is specified, request effect descriptor 2092 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2093 if (lStatus < 0) { 2094 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2095 goto Exit; 2096 } 2097 } else { 2098 // if uuid is not specified, look for an available implementation 2099 // of the required type in effect factory 2100 if (EffectIsNullUuid(&pDesc->type)) { 2101 ALOGW("createEffect() no effect type"); 2102 lStatus = BAD_VALUE; 2103 goto Exit; 2104 } 2105 uint32_t numEffects = 0; 2106 effect_descriptor_t d; 2107 d.flags = 0; // prevent compiler warning 2108 bool found = false; 2109 2110 lStatus = EffectQueryNumberEffects(&numEffects); 2111 if (lStatus < 0) { 2112 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2113 goto Exit; 2114 } 2115 for (uint32_t i = 0; i < numEffects; i++) { 2116 lStatus = EffectQueryEffect(i, &desc); 2117 if (lStatus < 0) { 2118 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2119 continue; 2120 } 2121 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2122 // If matching type found save effect descriptor. If the session is 2123 // 0 and the effect is not auxiliary, continue enumeration in case 2124 // an auxiliary version of this effect type is available 2125 found = true; 2126 d = desc; 2127 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2128 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2129 break; 2130 } 2131 } 2132 } 2133 if (!found) { 2134 lStatus = BAD_VALUE; 2135 ALOGW("createEffect() effect not found"); 2136 goto Exit; 2137 } 2138 // For same effect type, chose auxiliary version over insert version if 2139 // connect to output mix (Compliance to OpenSL ES) 2140 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2141 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2142 desc = d; 2143 } 2144 } 2145 2146 // Do not allow auxiliary effects on a session different from 0 (output mix) 2147 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2148 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2149 lStatus = INVALID_OPERATION; 2150 goto Exit; 2151 } 2152 2153 // check recording permission for visualizer 2154 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2155 !recordingAllowed()) { 2156 lStatus = PERMISSION_DENIED; 2157 goto Exit; 2158 } 2159 2160 // return effect descriptor 2161 *pDesc = desc; 2162 2163 // If output is not specified try to find a matching audio session ID in one of the 2164 // output threads. 2165 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2166 // because of code checking output when entering the function. 2167 // Note: io is never 0 when creating an effect on an input 2168 if (io == 0) { 2169 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2170 // output must be specified by AudioPolicyManager when using session 2171 // AUDIO_SESSION_OUTPUT_STAGE 2172 lStatus = BAD_VALUE; 2173 goto Exit; 2174 } 2175 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2176 // if the output returned by getOutputForEffect() is removed before we lock the 2177 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2178 // and we will exit safely 2179 io = AudioSystem::getOutputForEffect(&desc); 2180 ALOGV("createEffect got output %d", io); 2181 } 2182 if (io == 0) { 2183 // look for the thread where the specified audio session is present 2184 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2185 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2186 io = mPlaybackThreads.keyAt(i); 2187 break; 2188 } 2189 } 2190 if (io == 0) { 2191 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2192 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2193 io = mRecordThreads.keyAt(i); 2194 break; 2195 } 2196 } 2197 } 2198 } 2199 // If no output thread contains the requested session ID, default to 2200 // first output. The effect chain will be moved to the correct output 2201 // thread when a track with the same session ID is created 2202 if (io == 0 && mPlaybackThreads.size()) { 2203 io = mPlaybackThreads.keyAt(0); 2204 } 2205 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2206 } 2207 ThreadBase *thread = checkRecordThread_l(io); 2208 if (thread == NULL) { 2209 thread = checkPlaybackThread_l(io); 2210 if (thread == NULL) { 2211 ALOGE("createEffect() unknown output thread"); 2212 lStatus = BAD_VALUE; 2213 goto Exit; 2214 } 2215 } 2216 2217 sp<Client> client = registerPid_l(pid); 2218 2219 // create effect on selected output thread 2220 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2221 &desc, enabled, &lStatus); 2222 if (handle != 0 && id != NULL) { 2223 *id = handle->id(); 2224 } 2225 } 2226 2227Exit: 2228 if (status != NULL) { 2229 *status = lStatus; 2230 } 2231 return handle; 2232} 2233 2234status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2235 audio_io_handle_t dstOutput) 2236{ 2237 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2238 sessionId, srcOutput, dstOutput); 2239 Mutex::Autolock _l(mLock); 2240 if (srcOutput == dstOutput) { 2241 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2242 return NO_ERROR; 2243 } 2244 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2245 if (srcThread == NULL) { 2246 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2247 return BAD_VALUE; 2248 } 2249 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2250 if (dstThread == NULL) { 2251 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2252 return BAD_VALUE; 2253 } 2254 2255 Mutex::Autolock _dl(dstThread->mLock); 2256 Mutex::Autolock _sl(srcThread->mLock); 2257 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2258} 2259 2260// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2261status_t AudioFlinger::moveEffectChain_l(int sessionId, 2262 AudioFlinger::PlaybackThread *srcThread, 2263 AudioFlinger::PlaybackThread *dstThread, 2264 bool reRegister) 2265{ 2266 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2267 sessionId, srcThread, dstThread); 2268 2269 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2270 if (chain == 0) { 2271 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2272 sessionId, srcThread); 2273 return INVALID_OPERATION; 2274 } 2275 2276 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2277 // so that a new chain is created with correct parameters when first effect is added. This is 2278 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2279 // removed. 2280 srcThread->removeEffectChain_l(chain); 2281 2282 // transfer all effects one by one so that new effect chain is created on new thread with 2283 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2284 sp<EffectChain> dstChain; 2285 uint32_t strategy = 0; // prevent compiler warning 2286 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2287 Vector< sp<EffectModule> > removed; 2288 status_t status = NO_ERROR; 2289 while (effect != 0) { 2290 srcThread->removeEffect_l(effect); 2291 removed.add(effect); 2292 status = dstThread->addEffect_l(effect); 2293 if (status != NO_ERROR) { 2294 break; 2295 } 2296 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2297 if (effect->state() == EffectModule::ACTIVE || 2298 effect->state() == EffectModule::STOPPING) { 2299 effect->start(); 2300 } 2301 // if the move request is not received from audio policy manager, the effect must be 2302 // re-registered with the new strategy and output 2303 if (dstChain == 0) { 2304 dstChain = effect->chain().promote(); 2305 if (dstChain == 0) { 2306 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2307 status = NO_INIT; 2308 break; 2309 } 2310 strategy = dstChain->strategy(); 2311 } 2312 if (reRegister) { 2313 AudioSystem::unregisterEffect(effect->id()); 2314 AudioSystem::registerEffect(&effect->desc(), 2315 dstThread->id(), 2316 strategy, 2317 sessionId, 2318 effect->id()); 2319 } 2320 effect = chain->getEffectFromId_l(0); 2321 } 2322 2323 if (status != NO_ERROR) { 2324 for (size_t i = 0; i < removed.size(); i++) { 2325 srcThread->addEffect_l(removed[i]); 2326 if (dstChain != 0 && reRegister) { 2327 AudioSystem::unregisterEffect(removed[i]->id()); 2328 AudioSystem::registerEffect(&removed[i]->desc(), 2329 srcThread->id(), 2330 strategy, 2331 sessionId, 2332 removed[i]->id()); 2333 } 2334 } 2335 } 2336 2337 return status; 2338} 2339 2340bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2341{ 2342 if (mGlobalEffectEnableTime != 0 && 2343 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2344 return true; 2345 } 2346 2347 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2348 sp<EffectChain> ec = 2349 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2350 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2351 return true; 2352 } 2353 } 2354 return false; 2355} 2356 2357void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2358{ 2359 Mutex::Autolock _l(mLock); 2360 2361 mGlobalEffectEnableTime = systemTime(); 2362 2363 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2364 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2365 if (t->mType == ThreadBase::OFFLOAD) { 2366 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2367 } 2368 } 2369 2370} 2371 2372struct Entry { 2373#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2374 char mName[MAX_NAME]; 2375}; 2376 2377int comparEntry(const void *p1, const void *p2) 2378{ 2379 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2380} 2381 2382#ifdef TEE_SINK 2383void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2384{ 2385 NBAIO_Source *teeSource = source.get(); 2386 if (teeSource != NULL) { 2387 // .wav rotation 2388 // There is a benign race condition if 2 threads call this simultaneously. 2389 // They would both traverse the directory, but the result would simply be 2390 // failures at unlink() which are ignored. It's also unlikely since 2391 // normally dumpsys is only done by bugreport or from the command line. 2392 char teePath[32+256]; 2393 strcpy(teePath, "/data/misc/media"); 2394 size_t teePathLen = strlen(teePath); 2395 DIR *dir = opendir(teePath); 2396 teePath[teePathLen++] = '/'; 2397 if (dir != NULL) { 2398#define MAX_SORT 20 // number of entries to sort 2399#define MAX_KEEP 10 // number of entries to keep 2400 struct Entry entries[MAX_SORT]; 2401 size_t entryCount = 0; 2402 while (entryCount < MAX_SORT) { 2403 struct dirent de; 2404 struct dirent *result = NULL; 2405 int rc = readdir_r(dir, &de, &result); 2406 if (rc != 0) { 2407 ALOGW("readdir_r failed %d", rc); 2408 break; 2409 } 2410 if (result == NULL) { 2411 break; 2412 } 2413 if (result != &de) { 2414 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2415 break; 2416 } 2417 // ignore non .wav file entries 2418 size_t nameLen = strlen(de.d_name); 2419 if (nameLen <= 4 || nameLen >= MAX_NAME || 2420 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2421 continue; 2422 } 2423 strcpy(entries[entryCount++].mName, de.d_name); 2424 } 2425 (void) closedir(dir); 2426 if (entryCount > MAX_KEEP) { 2427 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2428 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2429 strcpy(&teePath[teePathLen], entries[i].mName); 2430 (void) unlink(teePath); 2431 } 2432 } 2433 } else { 2434 if (fd >= 0) { 2435 fdprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2436 } 2437 } 2438 char teeTime[16]; 2439 struct timeval tv; 2440 gettimeofday(&tv, NULL); 2441 struct tm tm; 2442 localtime_r(&tv.tv_sec, &tm); 2443 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2444 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2445 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2446 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2447 if (teeFd >= 0) { 2448 char wavHeader[44]; 2449 memcpy(wavHeader, 2450 "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", 2451 sizeof(wavHeader)); 2452 NBAIO_Format format = teeSource->format(); 2453 unsigned channelCount = Format_channelCount(format); 2454 ALOG_ASSERT(channelCount <= FCC_2); 2455 uint32_t sampleRate = Format_sampleRate(format); 2456 wavHeader[22] = channelCount; // number of channels 2457 wavHeader[24] = sampleRate; // sample rate 2458 wavHeader[25] = sampleRate >> 8; 2459 wavHeader[32] = channelCount * 2; // block alignment 2460 write(teeFd, wavHeader, sizeof(wavHeader)); 2461 size_t total = 0; 2462 bool firstRead = true; 2463 for (;;) { 2464#define TEE_SINK_READ 1024 2465 short buffer[TEE_SINK_READ * FCC_2]; 2466 size_t count = TEE_SINK_READ; 2467 ssize_t actual = teeSource->read(buffer, count, 2468 AudioBufferProvider::kInvalidPTS); 2469 bool wasFirstRead = firstRead; 2470 firstRead = false; 2471 if (actual <= 0) { 2472 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2473 continue; 2474 } 2475 break; 2476 } 2477 ALOG_ASSERT(actual <= (ssize_t)count); 2478 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2479 total += actual; 2480 } 2481 lseek(teeFd, (off_t) 4, SEEK_SET); 2482 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2483 write(teeFd, &temp, sizeof(temp)); 2484 lseek(teeFd, (off_t) 40, SEEK_SET); 2485 temp = total * channelCount * sizeof(short); 2486 write(teeFd, &temp, sizeof(temp)); 2487 close(teeFd); 2488 if (fd >= 0) { 2489 fdprintf(fd, "tee copied to %s\n", teePath); 2490 } 2491 } else { 2492 if (fd >= 0) { 2493 fdprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2494 } 2495 } 2496 } 2497} 2498#endif 2499 2500// ---------------------------------------------------------------------------- 2501 2502status_t AudioFlinger::onTransact( 2503 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2504{ 2505 return BnAudioFlinger::onTransact(code, data, reply, flags); 2506} 2507 2508}; // namespace android 2509