AudioFlinger.cpp revision 2829edccd7d2bb8244246f316face82b650b8949
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"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch (format & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump all hardware devs 422 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 423 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 424 dev->dump(dev, fd); 425 } 426 427#ifdef TEE_SINK 428 // dump the serially shared record tee sink 429 if (mRecordTeeSource != 0) { 430 dumpTee(fd, mRecordTeeSource); 431 } 432#endif 433 434 if (locked) { 435 mLock.unlock(); 436 } 437 438 // append a copy of media.log here by forwarding fd to it, but don't attempt 439 // to lookup the service if it's not running, as it will block for a second 440 if (mLogMemoryDealer != 0) { 441 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 442 if (binder != 0) { 443 dprintf(fd, "\nmedia.log:\n"); 444 Vector<String16> args; 445 binder->dump(fd, args); 446 } 447 } 448 } 449 return NO_ERROR; 450} 451 452sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 453{ 454 Mutex::Autolock _cl(mClientLock); 455 // If pid is already in the mClients wp<> map, then use that entry 456 // (for which promote() is always != 0), otherwise create a new entry and Client. 457 sp<Client> client = mClients.valueFor(pid).promote(); 458 if (client == 0) { 459 client = new Client(this, pid); 460 mClients.add(pid, client); 461 } 462 463 return client; 464} 465 466sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 467{ 468 // If there is no memory allocated for logs, return a dummy writer that does nothing 469 if (mLogMemoryDealer == 0) { 470 return new NBLog::Writer(); 471 } 472 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 473 // Similarly if we can't contact the media.log service, also return a dummy writer 474 if (binder == 0) { 475 return new NBLog::Writer(); 476 } 477 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 478 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 479 // If allocation fails, consult the vector of previously unregistered writers 480 // and garbage-collect one or more them until an allocation succeeds 481 if (shared == 0) { 482 Mutex::Autolock _l(mUnregisteredWritersLock); 483 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 484 { 485 // Pick the oldest stale writer to garbage-collect 486 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 487 mUnregisteredWriters.removeAt(0); 488 mediaLogService->unregisterWriter(iMemory); 489 // Now the media.log remote reference to IMemory is gone. When our last local 490 // reference to IMemory also drops to zero at end of this block, 491 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 492 } 493 // Re-attempt the allocation 494 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 495 if (shared != 0) { 496 goto success; 497 } 498 } 499 // Even after garbage-collecting all old writers, there is still not enough memory, 500 // so return a dummy writer 501 return new NBLog::Writer(); 502 } 503success: 504 mediaLogService->registerWriter(shared, size, name); 505 return new NBLog::Writer(size, shared); 506} 507 508void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 509{ 510 if (writer == 0) { 511 return; 512 } 513 sp<IMemory> iMemory(writer->getIMemory()); 514 if (iMemory == 0) { 515 return; 516 } 517 // Rather than removing the writer immediately, append it to a queue of old writers to 518 // be garbage-collected later. This allows us to continue to view old logs for a while. 519 Mutex::Autolock _l(mUnregisteredWritersLock); 520 mUnregisteredWriters.push(writer); 521} 522 523// IAudioFlinger interface 524 525 526sp<IAudioTrack> AudioFlinger::createTrack( 527 audio_stream_type_t streamType, 528 uint32_t sampleRate, 529 audio_format_t format, 530 audio_channel_mask_t channelMask, 531 size_t *frameCount, 532 IAudioFlinger::track_flags_t *flags, 533 const sp<IMemory>& sharedBuffer, 534 audio_io_handle_t output, 535 pid_t tid, 536 int *sessionId, 537 int clientUid, 538 status_t *status) 539{ 540 sp<PlaybackThread::Track> track; 541 sp<TrackHandle> trackHandle; 542 sp<Client> client; 543 status_t lStatus; 544 int lSessionId; 545 546 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 547 // but if someone uses binder directly they could bypass that and cause us to crash 548 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 549 ALOGE("createTrack() invalid stream type %d", streamType); 550 lStatus = BAD_VALUE; 551 goto Exit; 552 } 553 554 // further sample rate checks are performed by createTrack_l() depending on the thread type 555 if (sampleRate == 0) { 556 ALOGE("createTrack() invalid sample rate %u", sampleRate); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further channel mask checks are performed by createTrack_l() depending on the thread type 562 if (!audio_is_output_channel(channelMask)) { 563 ALOGE("createTrack() invalid channel mask %#x", channelMask); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further format checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_valid_format(format)) { 570 ALOGE("createTrack() invalid format %#x", format); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 576 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 { 582 Mutex::Autolock _l(mLock); 583 PlaybackThread *thread = checkPlaybackThread_l(output); 584 if (thread == NULL) { 585 ALOGE("no playback thread found for output handle %d", output); 586 lStatus = BAD_VALUE; 587 goto Exit; 588 } 589 590 pid_t pid = IPCThreadState::self()->getCallingPid(); 591 client = registerPid(pid); 592 593 PlaybackThread *effectThread = NULL; 594 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 595 lSessionId = *sessionId; 596 // check if an effect chain with the same session ID is present on another 597 // output thread and move it here. 598 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 599 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 600 if (mPlaybackThreads.keyAt(i) != output) { 601 uint32_t sessions = t->hasAudioSession(lSessionId); 602 if (sessions & PlaybackThread::EFFECT_SESSION) { 603 effectThread = t.get(); 604 break; 605 } 606 } 607 } 608 } else { 609 // if no audio session id is provided, create one here 610 lSessionId = nextUniqueId(); 611 if (sessionId != NULL) { 612 *sessionId = lSessionId; 613 } 614 } 615 ALOGV("createTrack() lSessionId: %d", lSessionId); 616 617 track = thread->createTrack_l(client, streamType, sampleRate, format, 618 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 619 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 620 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 621 622 // move effect chain to this output thread if an effect on same session was waiting 623 // for a track to be created 624 if (lStatus == NO_ERROR && effectThread != NULL) { 625 // no risk of deadlock because AudioFlinger::mLock is held 626 Mutex::Autolock _dl(thread->mLock); 627 Mutex::Autolock _sl(effectThread->mLock); 628 moveEffectChain_l(lSessionId, effectThread, thread, true); 629 } 630 631 // Look for sync events awaiting for a session to be used. 632 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 633 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 634 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 635 if (lStatus == NO_ERROR) { 636 (void) track->setSyncEvent(mPendingSyncEvents[i]); 637 } else { 638 mPendingSyncEvents[i]->cancel(); 639 } 640 mPendingSyncEvents.removeAt(i); 641 i--; 642 } 643 } 644 } 645 646 } 647 648 if (lStatus != NO_ERROR) { 649 // remove local strong reference to Client before deleting the Track so that the 650 // Client destructor is called by the TrackBase destructor with mClientLock held 651 // Don't hold mClientLock when releasing the reference on the track as the 652 // destructor will acquire it. 653 { 654 Mutex::Autolock _cl(mClientLock); 655 client.clear(); 656 } 657 track.clear(); 658 goto Exit; 659 } 660 661 // return handle to client 662 trackHandle = new TrackHandle(track); 663 664Exit: 665 *status = lStatus; 666 return trackHandle; 667} 668 669uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 670{ 671 Mutex::Autolock _l(mLock); 672 PlaybackThread *thread = checkPlaybackThread_l(output); 673 if (thread == NULL) { 674 ALOGW("sampleRate() unknown thread %d", output); 675 return 0; 676 } 677 return thread->sampleRate(); 678} 679 680audio_format_t AudioFlinger::format(audio_io_handle_t output) const 681{ 682 Mutex::Autolock _l(mLock); 683 PlaybackThread *thread = checkPlaybackThread_l(output); 684 if (thread == NULL) { 685 ALOGW("format() unknown thread %d", output); 686 return AUDIO_FORMAT_INVALID; 687 } 688 return thread->format(); 689} 690 691size_t AudioFlinger::frameCount(audio_io_handle_t output) const 692{ 693 Mutex::Autolock _l(mLock); 694 PlaybackThread *thread = checkPlaybackThread_l(output); 695 if (thread == NULL) { 696 ALOGW("frameCount() unknown thread %d", output); 697 return 0; 698 } 699 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 700 // should examine all callers and fix them to handle smaller counts 701 return thread->frameCount(); 702} 703 704uint32_t AudioFlinger::latency(audio_io_handle_t output) const 705{ 706 Mutex::Autolock _l(mLock); 707 PlaybackThread *thread = checkPlaybackThread_l(output); 708 if (thread == NULL) { 709 ALOGW("latency(): no playback thread found for output handle %d", output); 710 return 0; 711 } 712 return thread->latency(); 713} 714 715status_t AudioFlinger::setMasterVolume(float value) 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 mMasterVolume = value; 729 730 // Set master volume 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_VOLUME; 736 if (dev->canSetMasterVolume()) { 737 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 738 } 739 mHardwareStatus = AUDIO_HW_IDLE; 740 } 741 742 // Now set the master volume in each playback thread. Playback threads 743 // assigned to HALs which do not have master volume support will apply 744 // master volume during the mix operation. Threads with HALs which do 745 // support master volume will simply ignore the setting. 746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 747 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 748 749 return NO_ERROR; 750} 751 752status_t AudioFlinger::setMode(audio_mode_t mode) 753{ 754 status_t ret = initCheck(); 755 if (ret != NO_ERROR) { 756 return ret; 757 } 758 759 // check calling permissions 760 if (!settingsAllowed()) { 761 return PERMISSION_DENIED; 762 } 763 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 764 ALOGW("Illegal value: setMode(%d)", mode); 765 return BAD_VALUE; 766 } 767 768 { // scope for the lock 769 AutoMutex lock(mHardwareLock); 770 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 771 mHardwareStatus = AUDIO_HW_SET_MODE; 772 ret = dev->set_mode(dev, mode); 773 mHardwareStatus = AUDIO_HW_IDLE; 774 } 775 776 if (NO_ERROR == ret) { 777 Mutex::Autolock _l(mLock); 778 mMode = mode; 779 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 780 mPlaybackThreads.valueAt(i)->setMode(mode); 781 } 782 783 return ret; 784} 785 786status_t AudioFlinger::setMicMute(bool state) 787{ 788 status_t ret = initCheck(); 789 if (ret != NO_ERROR) { 790 return ret; 791 } 792 793 // check calling permissions 794 if (!settingsAllowed()) { 795 return PERMISSION_DENIED; 796 } 797 798 AutoMutex lock(mHardwareLock); 799 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 800 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 801 ret = dev->set_mic_mute(dev, state); 802 mHardwareStatus = AUDIO_HW_IDLE; 803 return ret; 804} 805 806bool AudioFlinger::getMicMute() const 807{ 808 status_t ret = initCheck(); 809 if (ret != NO_ERROR) { 810 return false; 811 } 812 813 bool state = AUDIO_MODE_INVALID; 814 AutoMutex lock(mHardwareLock); 815 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 816 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 817 dev->get_mic_mute(dev, &state); 818 mHardwareStatus = AUDIO_HW_IDLE; 819 return state; 820} 821 822status_t AudioFlinger::setMasterMute(bool muted) 823{ 824 status_t ret = initCheck(); 825 if (ret != NO_ERROR) { 826 return ret; 827 } 828 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 Mutex::Autolock _l(mLock); 835 mMasterMute = muted; 836 837 // Set master mute in the HALs which support it. 838 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 839 AutoMutex lock(mHardwareLock); 840 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 841 842 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 843 if (dev->canSetMasterMute()) { 844 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 845 } 846 mHardwareStatus = AUDIO_HW_IDLE; 847 } 848 849 // Now set the master mute in each playback thread. Playback threads 850 // assigned to HALs which do not have master mute support will apply master 851 // mute during the mix operation. Threads with HALs which do support master 852 // mute will simply ignore the setting. 853 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 854 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 855 856 return NO_ERROR; 857} 858 859float AudioFlinger::masterVolume() const 860{ 861 Mutex::Autolock _l(mLock); 862 return masterVolume_l(); 863} 864 865bool AudioFlinger::masterMute() const 866{ 867 Mutex::Autolock _l(mLock); 868 return masterMute_l(); 869} 870 871float AudioFlinger::masterVolume_l() const 872{ 873 return mMasterVolume; 874} 875 876bool AudioFlinger::masterMute_l() const 877{ 878 return mMasterMute; 879} 880 881status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 882 audio_io_handle_t output) 883{ 884 // check calling permissions 885 if (!settingsAllowed()) { 886 return PERMISSION_DENIED; 887 } 888 889 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 890 ALOGE("setStreamVolume() invalid stream %d", stream); 891 return BAD_VALUE; 892 } 893 894 AutoMutex lock(mLock); 895 PlaybackThread *thread = NULL; 896 if (output != AUDIO_IO_HANDLE_NONE) { 897 thread = checkPlaybackThread_l(output); 898 if (thread == NULL) { 899 return BAD_VALUE; 900 } 901 } 902 903 mStreamTypes[stream].volume = value; 904 905 if (thread == NULL) { 906 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 907 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 908 } 909 } else { 910 thread->setStreamVolume(stream, value); 911 } 912 913 return NO_ERROR; 914} 915 916status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 917{ 918 // check calling permissions 919 if (!settingsAllowed()) { 920 return PERMISSION_DENIED; 921 } 922 923 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 924 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 925 ALOGE("setStreamMute() invalid stream %d", stream); 926 return BAD_VALUE; 927 } 928 929 AutoMutex lock(mLock); 930 mStreamTypes[stream].mute = muted; 931 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 932 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 933 934 return NO_ERROR; 935} 936 937float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 938{ 939 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 940 return 0.0f; 941 } 942 943 AutoMutex lock(mLock); 944 float volume; 945 if (output != AUDIO_IO_HANDLE_NONE) { 946 PlaybackThread *thread = checkPlaybackThread_l(output); 947 if (thread == NULL) { 948 return 0.0f; 949 } 950 volume = thread->streamVolume(stream); 951 } else { 952 volume = streamVolume_l(stream); 953 } 954 955 return volume; 956} 957 958bool AudioFlinger::streamMute(audio_stream_type_t stream) const 959{ 960 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 961 return true; 962 } 963 964 AutoMutex lock(mLock); 965 return streamMute_l(stream); 966} 967 968status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 969{ 970 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 971 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 972 973 // check calling permissions 974 if (!settingsAllowed()) { 975 return PERMISSION_DENIED; 976 } 977 978 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 979 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 980 Mutex::Autolock _l(mLock); 981 status_t final_result = NO_ERROR; 982 { 983 AutoMutex lock(mHardwareLock); 984 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 985 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 986 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 987 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 988 final_result = result ?: final_result; 989 } 990 mHardwareStatus = AUDIO_HW_IDLE; 991 } 992 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 993 AudioParameter param = AudioParameter(keyValuePairs); 994 String8 value; 995 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 996 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 997 if (mBtNrecIsOff != btNrecIsOff) { 998 for (size_t i = 0; i < mRecordThreads.size(); i++) { 999 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1000 audio_devices_t device = thread->inDevice(); 1001 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1002 // collect all of the thread's session IDs 1003 KeyedVector<int, bool> ids = thread->sessionIds(); 1004 // suspend effects associated with those session IDs 1005 for (size_t j = 0; j < ids.size(); ++j) { 1006 int sessionId = ids.keyAt(j); 1007 thread->setEffectSuspended(FX_IID_AEC, 1008 suspend, 1009 sessionId); 1010 thread->setEffectSuspended(FX_IID_NS, 1011 suspend, 1012 sessionId); 1013 } 1014 } 1015 mBtNrecIsOff = btNrecIsOff; 1016 } 1017 } 1018 String8 screenState; 1019 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1020 bool isOff = screenState == "off"; 1021 if (isOff != (AudioFlinger::mScreenState & 1)) { 1022 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1023 } 1024 } 1025 return final_result; 1026 } 1027 1028 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1029 // and the thread is exited once the lock is released 1030 sp<ThreadBase> thread; 1031 { 1032 Mutex::Autolock _l(mLock); 1033 thread = checkPlaybackThread_l(ioHandle); 1034 if (thread == 0) { 1035 thread = checkRecordThread_l(ioHandle); 1036 } else if (thread == primaryPlaybackThread_l()) { 1037 // indicate output device change to all input threads for pre processing 1038 AudioParameter param = AudioParameter(keyValuePairs); 1039 int value; 1040 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1041 (value != 0)) { 1042 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1043 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1044 } 1045 } 1046 } 1047 } 1048 if (thread != 0) { 1049 return thread->setParameters(keyValuePairs); 1050 } 1051 return BAD_VALUE; 1052} 1053 1054String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1055{ 1056 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1057 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1058 1059 Mutex::Autolock _l(mLock); 1060 1061 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1062 String8 out_s8; 1063 1064 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1065 char *s; 1066 { 1067 AutoMutex lock(mHardwareLock); 1068 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1069 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1070 s = dev->get_parameters(dev, keys.string()); 1071 mHardwareStatus = AUDIO_HW_IDLE; 1072 } 1073 out_s8 += String8(s ? s : ""); 1074 free(s); 1075 } 1076 return out_s8; 1077 } 1078 1079 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1080 if (playbackThread != NULL) { 1081 return playbackThread->getParameters(keys); 1082 } 1083 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1084 if (recordThread != NULL) { 1085 return recordThread->getParameters(keys); 1086 } 1087 return String8(""); 1088} 1089 1090size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1091 audio_channel_mask_t channelMask) const 1092{ 1093 status_t ret = initCheck(); 1094 if (ret != NO_ERROR) { 1095 return 0; 1096 } 1097 1098 AutoMutex lock(mHardwareLock); 1099 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1100 struct audio_config config; 1101 memset(&config, 0, sizeof(config)); 1102 config.sample_rate = sampleRate; 1103 config.channel_mask = channelMask; 1104 config.format = format; 1105 1106 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1107 size_t size = dev->get_input_buffer_size(dev, &config); 1108 mHardwareStatus = AUDIO_HW_IDLE; 1109 return size; 1110} 1111 1112uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1113{ 1114 Mutex::Autolock _l(mLock); 1115 1116 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1117 if (recordThread != NULL) { 1118 return recordThread->getInputFramesLost(); 1119 } 1120 return 0; 1121} 1122 1123status_t AudioFlinger::setVoiceVolume(float value) 1124{ 1125 status_t ret = initCheck(); 1126 if (ret != NO_ERROR) { 1127 return ret; 1128 } 1129 1130 // check calling permissions 1131 if (!settingsAllowed()) { 1132 return PERMISSION_DENIED; 1133 } 1134 1135 AutoMutex lock(mHardwareLock); 1136 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1137 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1138 ret = dev->set_voice_volume(dev, value); 1139 mHardwareStatus = AUDIO_HW_IDLE; 1140 1141 return ret; 1142} 1143 1144status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1145 audio_io_handle_t output) const 1146{ 1147 status_t status; 1148 1149 Mutex::Autolock _l(mLock); 1150 1151 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1152 if (playbackThread != NULL) { 1153 return playbackThread->getRenderPosition(halFrames, dspFrames); 1154 } 1155 1156 return BAD_VALUE; 1157} 1158 1159void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1160{ 1161 Mutex::Autolock _l(mLock); 1162 bool clientAdded = false; 1163 { 1164 Mutex::Autolock _cl(mClientLock); 1165 1166 pid_t pid = IPCThreadState::self()->getCallingPid(); 1167 if (mNotificationClients.indexOfKey(pid) < 0) { 1168 sp<NotificationClient> notificationClient = new NotificationClient(this, 1169 client, 1170 pid); 1171 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1172 1173 mNotificationClients.add(pid, notificationClient); 1174 1175 sp<IBinder> binder = client->asBinder(); 1176 binder->linkToDeath(notificationClient); 1177 clientAdded = true; 1178 } 1179 } 1180 1181 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1182 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1183 if (clientAdded) { 1184 // the config change is always sent from playback or record threads to avoid deadlock 1185 // with AudioSystem::gLock 1186 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1187 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1188 } 1189 1190 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1191 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1192 } 1193 } 1194} 1195 1196void AudioFlinger::removeNotificationClient(pid_t pid) 1197{ 1198 Mutex::Autolock _l(mLock); 1199 { 1200 Mutex::Autolock _cl(mClientLock); 1201 mNotificationClients.removeItem(pid); 1202 } 1203 1204 ALOGV("%d died, releasing its sessions", pid); 1205 size_t num = mAudioSessionRefs.size(); 1206 bool removed = false; 1207 for (size_t i = 0; i< num; ) { 1208 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1209 ALOGV(" pid %d @ %d", ref->mPid, i); 1210 if (ref->mPid == pid) { 1211 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1212 mAudioSessionRefs.removeAt(i); 1213 delete ref; 1214 removed = true; 1215 num--; 1216 } else { 1217 i++; 1218 } 1219 } 1220 if (removed) { 1221 purgeStaleEffects_l(); 1222 } 1223} 1224 1225void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1226{ 1227 Mutex::Autolock _l(mClientLock); 1228 size_t size = mNotificationClients.size(); 1229 for (size_t i = 0; i < size; i++) { 1230 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1231 ioHandle, 1232 param2); 1233 } 1234} 1235 1236// removeClient_l() must be called with AudioFlinger::mClientLock held 1237void AudioFlinger::removeClient_l(pid_t pid) 1238{ 1239 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1240 IPCThreadState::self()->getCallingPid()); 1241 mClients.removeItem(pid); 1242} 1243 1244// getEffectThread_l() must be called with AudioFlinger::mLock held 1245sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1246{ 1247 sp<PlaybackThread> thread; 1248 1249 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1250 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1251 ALOG_ASSERT(thread == 0); 1252 thread = mPlaybackThreads.valueAt(i); 1253 } 1254 } 1255 1256 return thread; 1257} 1258 1259 1260 1261// ---------------------------------------------------------------------------- 1262 1263AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1264 : RefBase(), 1265 mAudioFlinger(audioFlinger), 1266 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1267 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1268 mPid(pid), 1269 mTimedTrackCount(0) 1270{ 1271 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1272} 1273 1274// Client destructor must be called with AudioFlinger::mClientLock held 1275AudioFlinger::Client::~Client() 1276{ 1277 mAudioFlinger->removeClient_l(mPid); 1278} 1279 1280sp<MemoryDealer> AudioFlinger::Client::heap() const 1281{ 1282 return mMemoryDealer; 1283} 1284 1285// Reserve one of the limited slots for a timed audio track associated 1286// with this client 1287bool AudioFlinger::Client::reserveTimedTrack() 1288{ 1289 const int kMaxTimedTracksPerClient = 4; 1290 1291 Mutex::Autolock _l(mTimedTrackLock); 1292 1293 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1294 ALOGW("can not create timed track - pid %d has exceeded the limit", 1295 mPid); 1296 return false; 1297 } 1298 1299 mTimedTrackCount++; 1300 return true; 1301} 1302 1303// Release a slot for a timed audio track 1304void AudioFlinger::Client::releaseTimedTrack() 1305{ 1306 Mutex::Autolock _l(mTimedTrackLock); 1307 mTimedTrackCount--; 1308} 1309 1310// ---------------------------------------------------------------------------- 1311 1312AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1313 const sp<IAudioFlingerClient>& client, 1314 pid_t pid) 1315 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1316{ 1317} 1318 1319AudioFlinger::NotificationClient::~NotificationClient() 1320{ 1321} 1322 1323void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1324{ 1325 sp<NotificationClient> keep(this); 1326 mAudioFlinger->removeNotificationClient(mPid); 1327} 1328 1329 1330// ---------------------------------------------------------------------------- 1331 1332static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1333 return audio_is_remote_submix_device(inDevice); 1334} 1335 1336sp<IAudioRecord> AudioFlinger::openRecord( 1337 audio_io_handle_t input, 1338 uint32_t sampleRate, 1339 audio_format_t format, 1340 audio_channel_mask_t channelMask, 1341 size_t *frameCount, 1342 IAudioFlinger::track_flags_t *flags, 1343 pid_t tid, 1344 int *sessionId, 1345 size_t *notificationFrames, 1346 sp<IMemory>& cblk, 1347 sp<IMemory>& buffers, 1348 status_t *status) 1349{ 1350 sp<RecordThread::RecordTrack> recordTrack; 1351 sp<RecordHandle> recordHandle; 1352 sp<Client> client; 1353 status_t lStatus; 1354 int lSessionId; 1355 1356 cblk.clear(); 1357 buffers.clear(); 1358 1359 // check calling permissions 1360 if (!recordingAllowed()) { 1361 ALOGE("openRecord() permission denied: recording not allowed"); 1362 lStatus = PERMISSION_DENIED; 1363 goto Exit; 1364 } 1365 1366 // further sample rate checks are performed by createRecordTrack_l() 1367 if (sampleRate == 0) { 1368 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1369 lStatus = BAD_VALUE; 1370 goto Exit; 1371 } 1372 1373 // we don't yet support anything other than 16-bit PCM 1374 if (!(audio_is_valid_format(format) && 1375 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1376 ALOGE("openRecord() invalid format %#x", format); 1377 lStatus = BAD_VALUE; 1378 goto Exit; 1379 } 1380 1381 // further channel mask checks are performed by createRecordTrack_l() 1382 if (!audio_is_input_channel(channelMask)) { 1383 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1384 lStatus = BAD_VALUE; 1385 goto Exit; 1386 } 1387 1388 { 1389 Mutex::Autolock _l(mLock); 1390 RecordThread *thread = checkRecordThread_l(input); 1391 if (thread == NULL) { 1392 ALOGE("openRecord() checkRecordThread_l failed"); 1393 lStatus = BAD_VALUE; 1394 goto Exit; 1395 } 1396 1397 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1398 && !captureAudioOutputAllowed()) { 1399 ALOGE("openRecord() permission denied: capture not allowed"); 1400 lStatus = PERMISSION_DENIED; 1401 goto Exit; 1402 } 1403 1404 pid_t pid = IPCThreadState::self()->getCallingPid(); 1405 client = registerPid(pid); 1406 1407 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1408 lSessionId = *sessionId; 1409 } else { 1410 // if no audio session id is provided, create one here 1411 lSessionId = nextUniqueId(); 1412 if (sessionId != NULL) { 1413 *sessionId = lSessionId; 1414 } 1415 } 1416 ALOGV("openRecord() lSessionId: %d", lSessionId); 1417 1418 // TODO: the uid should be passed in as a parameter to openRecord 1419 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1420 frameCount, lSessionId, notificationFrames, 1421 IPCThreadState::self()->getCallingUid(), 1422 flags, tid, &lStatus); 1423 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1424 } 1425 1426 if (lStatus != NO_ERROR) { 1427 // remove local strong reference to Client before deleting the RecordTrack so that the 1428 // Client destructor is called by the TrackBase destructor with mClientLock held 1429 // Don't hold mClientLock when releasing the reference on the track as the 1430 // destructor will acquire it. 1431 { 1432 Mutex::Autolock _cl(mClientLock); 1433 client.clear(); 1434 } 1435 recordTrack.clear(); 1436 goto Exit; 1437 } 1438 1439 cblk = recordTrack->getCblk(); 1440 buffers = recordTrack->getBuffers(); 1441 1442 // return handle to client 1443 recordHandle = new RecordHandle(recordTrack); 1444 1445Exit: 1446 *status = lStatus; 1447 return recordHandle; 1448} 1449 1450 1451 1452// ---------------------------------------------------------------------------- 1453 1454audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1455{ 1456 if (!settingsAllowed()) { 1457 return 0; 1458 } 1459 Mutex::Autolock _l(mLock); 1460 return loadHwModule_l(name); 1461} 1462 1463// loadHwModule_l() must be called with AudioFlinger::mLock held 1464audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1465{ 1466 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1467 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1468 ALOGW("loadHwModule() module %s already loaded", name); 1469 return mAudioHwDevs.keyAt(i); 1470 } 1471 } 1472 1473 audio_hw_device_t *dev; 1474 1475 int rc = load_audio_interface(name, &dev); 1476 if (rc) { 1477 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1478 return 0; 1479 } 1480 1481 mHardwareStatus = AUDIO_HW_INIT; 1482 rc = dev->init_check(dev); 1483 mHardwareStatus = AUDIO_HW_IDLE; 1484 if (rc) { 1485 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1486 return 0; 1487 } 1488 1489 // Check and cache this HAL's level of support for master mute and master 1490 // volume. If this is the first HAL opened, and it supports the get 1491 // methods, use the initial values provided by the HAL as the current 1492 // master mute and volume settings. 1493 1494 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1495 { // scope for auto-lock pattern 1496 AutoMutex lock(mHardwareLock); 1497 1498 if (0 == mAudioHwDevs.size()) { 1499 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1500 if (NULL != dev->get_master_volume) { 1501 float mv; 1502 if (OK == dev->get_master_volume(dev, &mv)) { 1503 mMasterVolume = mv; 1504 } 1505 } 1506 1507 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1508 if (NULL != dev->get_master_mute) { 1509 bool mm; 1510 if (OK == dev->get_master_mute(dev, &mm)) { 1511 mMasterMute = mm; 1512 } 1513 } 1514 } 1515 1516 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1517 if ((NULL != dev->set_master_volume) && 1518 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1519 flags = static_cast<AudioHwDevice::Flags>(flags | 1520 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1521 } 1522 1523 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1524 if ((NULL != dev->set_master_mute) && 1525 (OK == dev->set_master_mute(dev, mMasterMute))) { 1526 flags = static_cast<AudioHwDevice::Flags>(flags | 1527 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1528 } 1529 1530 mHardwareStatus = AUDIO_HW_IDLE; 1531 } 1532 1533 audio_module_handle_t handle = nextUniqueId(); 1534 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev, flags)); 1535 1536 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1537 name, dev->common.module->name, dev->common.module->id, handle); 1538 1539 return handle; 1540 1541} 1542 1543// ---------------------------------------------------------------------------- 1544 1545uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1546{ 1547 Mutex::Autolock _l(mLock); 1548 PlaybackThread *thread = primaryPlaybackThread_l(); 1549 return thread != NULL ? thread->sampleRate() : 0; 1550} 1551 1552size_t AudioFlinger::getPrimaryOutputFrameCount() 1553{ 1554 Mutex::Autolock _l(mLock); 1555 PlaybackThread *thread = primaryPlaybackThread_l(); 1556 return thread != NULL ? thread->frameCountHAL() : 0; 1557} 1558 1559// ---------------------------------------------------------------------------- 1560 1561status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1562{ 1563 uid_t uid = IPCThreadState::self()->getCallingUid(); 1564 if (uid != AID_SYSTEM) { 1565 return PERMISSION_DENIED; 1566 } 1567 Mutex::Autolock _l(mLock); 1568 if (mIsDeviceTypeKnown) { 1569 return INVALID_OPERATION; 1570 } 1571 mIsLowRamDevice = isLowRamDevice; 1572 mIsDeviceTypeKnown = true; 1573 return NO_ERROR; 1574} 1575 1576// ---------------------------------------------------------------------------- 1577 1578audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 1579 audio_devices_t *pDevices, 1580 uint32_t *pSamplingRate, 1581 audio_format_t *pFormat, 1582 audio_channel_mask_t *pChannelMask, 1583 uint32_t *pLatencyMs, 1584 audio_output_flags_t flags, 1585 const audio_offload_info_t *offloadInfo) 1586{ 1587 struct audio_config config; 1588 memset(&config, 0, sizeof(config)); 1589 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1590 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1591 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1592 if (offloadInfo != NULL) { 1593 config.offload_info = *offloadInfo; 1594 } 1595 1596 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1597 module, 1598 (pDevices != NULL) ? *pDevices : 0, 1599 config.sample_rate, 1600 config.format, 1601 config.channel_mask, 1602 flags); 1603 ALOGV("openOutput(), offloadInfo %p version 0x%04x", 1604 offloadInfo, offloadInfo == NULL ? -1 : offloadInfo->version); 1605 1606 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1607 return AUDIO_IO_HANDLE_NONE; 1608 } 1609 1610 Mutex::Autolock _l(mLock); 1611 1612 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, *pDevices); 1613 if (outHwDev == NULL) { 1614 return AUDIO_IO_HANDLE_NONE; 1615 } 1616 1617 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1618 audio_io_handle_t id = nextUniqueId(); 1619 1620 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1621 1622 audio_stream_out_t *outStream = NULL; 1623 1624 // FOR TESTING ONLY: 1625 // Enable increased sink precision for mixing mode if kEnableExtendedPrecision is true. 1626 if (kEnableExtendedPrecision && // Check only for Normal Mixing mode 1627 !(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1628 // Update format 1629 //config.format = AUDIO_FORMAT_PCM_FLOAT; 1630 //config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1631 //config.format = AUDIO_FORMAT_PCM_32_BIT; 1632 //config.format = AUDIO_FORMAT_PCM_8_24_BIT; 1633 // ALOGV("openOutput() upgrading format to %#08x", config.format); 1634 } 1635 1636 status_t status = hwDevHal->open_output_stream(hwDevHal, 1637 id, 1638 *pDevices, 1639 (audio_output_flags_t)flags, 1640 &config, 1641 &outStream); 1642 1643 mHardwareStatus = AUDIO_HW_IDLE; 1644 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %#08x, " 1645 "Channels %x, status %d", 1646 outStream, 1647 config.sample_rate, 1648 config.format, 1649 config.channel_mask, 1650 status); 1651 1652 if (status == NO_ERROR && outStream != NULL) { 1653 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream, flags); 1654 1655 PlaybackThread *thread; 1656 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1657 thread = new OffloadThread(this, output, id, *pDevices); 1658 ALOGV("openOutput() created offload output: ID %d thread %p", id, thread); 1659 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1660 || !isValidPcmSinkFormat(config.format) 1661 || (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 1662 thread = new DirectOutputThread(this, output, id, *pDevices); 1663 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 1664 } else { 1665 thread = new MixerThread(this, output, id, *pDevices); 1666 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 1667 } 1668 mPlaybackThreads.add(id, thread); 1669 1670 if (pSamplingRate != NULL) { 1671 *pSamplingRate = config.sample_rate; 1672 } 1673 if (pFormat != NULL) { 1674 *pFormat = config.format; 1675 } 1676 if (pChannelMask != NULL) { 1677 *pChannelMask = config.channel_mask; 1678 } 1679 if (pLatencyMs != NULL) { 1680 *pLatencyMs = thread->latency(); 1681 } 1682 1683 // notify client processes of the new output creation 1684 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1685 1686 // the first primary output opened designates the primary hw device 1687 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1688 ALOGI("Using module %d has the primary audio interface", module); 1689 mPrimaryHardwareDev = outHwDev; 1690 1691 AutoMutex lock(mHardwareLock); 1692 mHardwareStatus = AUDIO_HW_SET_MODE; 1693 hwDevHal->set_mode(hwDevHal, mMode); 1694 mHardwareStatus = AUDIO_HW_IDLE; 1695 1696 mPrimaryOutputSampleRate = config.sample_rate; 1697 } 1698 return id; 1699 } 1700 1701 return AUDIO_IO_HANDLE_NONE; 1702} 1703 1704audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1705 audio_io_handle_t output2) 1706{ 1707 Mutex::Autolock _l(mLock); 1708 MixerThread *thread1 = checkMixerThread_l(output1); 1709 MixerThread *thread2 = checkMixerThread_l(output2); 1710 1711 if (thread1 == NULL || thread2 == NULL) { 1712 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1713 output2); 1714 return AUDIO_IO_HANDLE_NONE; 1715 } 1716 1717 audio_io_handle_t id = nextUniqueId(); 1718 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1719 thread->addOutputTrack(thread2); 1720 mPlaybackThreads.add(id, thread); 1721 // notify client processes of the new output creation 1722 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1723 return id; 1724} 1725 1726status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1727{ 1728 return closeOutput_nonvirtual(output); 1729} 1730 1731status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1732{ 1733 // keep strong reference on the playback thread so that 1734 // it is not destroyed while exit() is executed 1735 sp<PlaybackThread> thread; 1736 { 1737 Mutex::Autolock _l(mLock); 1738 thread = checkPlaybackThread_l(output); 1739 if (thread == NULL) { 1740 return BAD_VALUE; 1741 } 1742 1743 ALOGV("closeOutput() %d", output); 1744 1745 if (thread->type() == ThreadBase::MIXER) { 1746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1747 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1748 DuplicatingThread *dupThread = 1749 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1750 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1751 1752 } 1753 } 1754 } 1755 1756 1757 mPlaybackThreads.removeItem(output); 1758 // save all effects to the default thread 1759 if (mPlaybackThreads.size()) { 1760 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1761 if (dstThread != NULL) { 1762 // audioflinger lock is held here so the acquisition order of thread locks does not 1763 // matter 1764 Mutex::Autolock _dl(dstThread->mLock); 1765 Mutex::Autolock _sl(thread->mLock); 1766 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1767 for (size_t i = 0; i < effectChains.size(); i ++) { 1768 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1769 } 1770 } 1771 } 1772 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1773 } 1774 thread->exit(); 1775 // The thread entity (active unit of execution) is no longer running here, 1776 // but the ThreadBase container still exists. 1777 1778 if (thread->type() != ThreadBase::DUPLICATING) { 1779 AudioStreamOut *out = thread->clearOutput(); 1780 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1781 // from now on thread->mOutput is NULL 1782 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1783 delete out; 1784 } 1785 return NO_ERROR; 1786} 1787 1788status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1789{ 1790 Mutex::Autolock _l(mLock); 1791 PlaybackThread *thread = checkPlaybackThread_l(output); 1792 1793 if (thread == NULL) { 1794 return BAD_VALUE; 1795 } 1796 1797 ALOGV("suspendOutput() %d", output); 1798 thread->suspend(); 1799 1800 return NO_ERROR; 1801} 1802 1803status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1804{ 1805 Mutex::Autolock _l(mLock); 1806 PlaybackThread *thread = checkPlaybackThread_l(output); 1807 1808 if (thread == NULL) { 1809 return BAD_VALUE; 1810 } 1811 1812 ALOGV("restoreOutput() %d", output); 1813 1814 thread->restore(); 1815 1816 return NO_ERROR; 1817} 1818 1819audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 1820 audio_devices_t *pDevices, 1821 uint32_t *pSamplingRate, 1822 audio_format_t *pFormat, 1823 audio_channel_mask_t *pChannelMask) 1824{ 1825 struct audio_config config; 1826 memset(&config, 0, sizeof(config)); 1827 config.sample_rate = (pSamplingRate != NULL) ? *pSamplingRate : 0; 1828 config.channel_mask = (pChannelMask != NULL) ? *pChannelMask : 0; 1829 config.format = (pFormat != NULL) ? *pFormat : AUDIO_FORMAT_DEFAULT; 1830 1831 uint32_t reqSamplingRate = config.sample_rate; 1832 audio_format_t reqFormat = config.format; 1833 audio_channel_mask_t reqChannelMask = config.channel_mask; 1834 1835 if (pDevices == NULL || *pDevices == AUDIO_DEVICE_NONE) { 1836 return 0; 1837 } 1838 1839 Mutex::Autolock _l(mLock); 1840 1841 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, *pDevices); 1842 if (inHwDev == NULL) { 1843 return 0; 1844 } 1845 1846 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1847 audio_io_handle_t id = nextUniqueId(); 1848 1849 audio_stream_in_t *inStream = NULL; 1850 status_t status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, 1851 &inStream); 1852 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %#x, Channels %x, " 1853 "status %d", 1854 inStream, 1855 config.sample_rate, 1856 config.format, 1857 config.channel_mask, 1858 status); 1859 1860 // If the input could not be opened with the requested parameters and we can handle the 1861 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1862 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1863 if (status == BAD_VALUE && 1864 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 1865 (config.sample_rate <= 2 * reqSamplingRate) && 1866 (audio_channel_count_from_in_mask(config.channel_mask) <= FCC_2) && 1867 (audio_channel_count_from_in_mask(reqChannelMask) <= FCC_2)) { 1868 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1869 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 1870 inStream = NULL; 1871 status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config, &inStream); 1872 // FIXME log this new status; HAL should not propose any further changes 1873 } 1874 1875 if (status == NO_ERROR && inStream != NULL) { 1876 1877#ifdef TEE_SINK 1878 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1879 // or (re-)create if current Pipe is idle and does not match the new format 1880 sp<NBAIO_Sink> teeSink; 1881 enum { 1882 TEE_SINK_NO, // don't copy input 1883 TEE_SINK_NEW, // copy input using a new pipe 1884 TEE_SINK_OLD, // copy input using an existing pipe 1885 } kind; 1886 NBAIO_Format format = Format_from_SR_C(inStream->common.get_sample_rate(&inStream->common), 1887 audio_channel_count_from_in_mask( 1888 inStream->common.get_channels(&inStream->common))); 1889 if (!mTeeSinkInputEnabled) { 1890 kind = TEE_SINK_NO; 1891 } else if (!Format_isValid(format)) { 1892 kind = TEE_SINK_NO; 1893 } else if (mRecordTeeSink == 0) { 1894 kind = TEE_SINK_NEW; 1895 } else if (mRecordTeeSink->getStrongCount() != 1) { 1896 kind = TEE_SINK_NO; 1897 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1898 kind = TEE_SINK_OLD; 1899 } else { 1900 kind = TEE_SINK_NEW; 1901 } 1902 switch (kind) { 1903 case TEE_SINK_NEW: { 1904 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1905 size_t numCounterOffers = 0; 1906 const NBAIO_Format offers[1] = {format}; 1907 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1908 ALOG_ASSERT(index == 0); 1909 PipeReader *pipeReader = new PipeReader(*pipe); 1910 numCounterOffers = 0; 1911 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1912 ALOG_ASSERT(index == 0); 1913 mRecordTeeSink = pipe; 1914 mRecordTeeSource = pipeReader; 1915 teeSink = pipe; 1916 } 1917 break; 1918 case TEE_SINK_OLD: 1919 teeSink = mRecordTeeSink; 1920 break; 1921 case TEE_SINK_NO: 1922 default: 1923 break; 1924 } 1925#endif 1926 1927 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 1928 1929 // Start record thread 1930 // RecordThread requires both input and output device indication to forward to audio 1931 // pre processing modules 1932 RecordThread *thread = new RecordThread(this, 1933 input, 1934 id, 1935 primaryOutputDevice_l(), 1936 *pDevices 1937#ifdef TEE_SINK 1938 , teeSink 1939#endif 1940 ); 1941 mRecordThreads.add(id, thread); 1942 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 1943 if (pSamplingRate != NULL) { 1944 *pSamplingRate = reqSamplingRate; 1945 } 1946 if (pFormat != NULL) { 1947 *pFormat = config.format; 1948 } 1949 if (pChannelMask != NULL) { 1950 *pChannelMask = reqChannelMask; 1951 } 1952 1953 // notify client processes of the new input creation 1954 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1955 return id; 1956 } 1957 1958 return 0; 1959} 1960 1961status_t AudioFlinger::closeInput(audio_io_handle_t input) 1962{ 1963 return closeInput_nonvirtual(input); 1964} 1965 1966status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 1967{ 1968 // keep strong reference on the record thread so that 1969 // it is not destroyed while exit() is executed 1970 sp<RecordThread> thread; 1971 { 1972 Mutex::Autolock _l(mLock); 1973 thread = checkRecordThread_l(input); 1974 if (thread == 0) { 1975 return BAD_VALUE; 1976 } 1977 1978 ALOGV("closeInput() %d", input); 1979 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 1980 mRecordThreads.removeItem(input); 1981 } 1982 thread->exit(); 1983 // The thread entity (active unit of execution) is no longer running here, 1984 // but the ThreadBase container still exists. 1985 1986 AudioStreamIn *in = thread->clearInput(); 1987 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 1988 // from now on thread->mInput is NULL 1989 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 1990 delete in; 1991 1992 return NO_ERROR; 1993} 1994 1995status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 1996{ 1997 Mutex::Autolock _l(mLock); 1998 ALOGV("invalidateStream() stream %d", stream); 1999 2000 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2001 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2002 thread->invalidateTracks(stream); 2003 } 2004 2005 return NO_ERROR; 2006} 2007 2008 2009int AudioFlinger::newAudioSessionId() 2010{ 2011 return nextUniqueId(); 2012} 2013 2014void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2015{ 2016 Mutex::Autolock _l(mLock); 2017 pid_t caller = IPCThreadState::self()->getCallingPid(); 2018 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2019 if (pid != -1 && (caller == getpid_cached)) { 2020 caller = pid; 2021 } 2022 2023 { 2024 Mutex::Autolock _cl(mClientLock); 2025 // Ignore requests received from processes not known as notification client. The request 2026 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2027 // called from a different pid leaving a stale session reference. Also we don't know how 2028 // to clear this reference if the client process dies. 2029 if (mNotificationClients.indexOfKey(caller) < 0) { 2030 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2031 return; 2032 } 2033 } 2034 2035 size_t num = mAudioSessionRefs.size(); 2036 for (size_t i = 0; i< num; i++) { 2037 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2038 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2039 ref->mCnt++; 2040 ALOGV(" incremented refcount to %d", ref->mCnt); 2041 return; 2042 } 2043 } 2044 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2045 ALOGV(" added new entry for %d", audioSession); 2046} 2047 2048void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2049{ 2050 Mutex::Autolock _l(mLock); 2051 pid_t caller = IPCThreadState::self()->getCallingPid(); 2052 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2053 if (pid != -1 && (caller == getpid_cached)) { 2054 caller = pid; 2055 } 2056 size_t num = mAudioSessionRefs.size(); 2057 for (size_t i = 0; i< num; i++) { 2058 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2059 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2060 ref->mCnt--; 2061 ALOGV(" decremented refcount to %d", ref->mCnt); 2062 if (ref->mCnt == 0) { 2063 mAudioSessionRefs.removeAt(i); 2064 delete ref; 2065 purgeStaleEffects_l(); 2066 } 2067 return; 2068 } 2069 } 2070 // If the caller is mediaserver it is likely that the session being released was acquired 2071 // on behalf of a process not in notification clients and we ignore the warning. 2072 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2073} 2074 2075void AudioFlinger::purgeStaleEffects_l() { 2076 2077 ALOGV("purging stale effects"); 2078 2079 Vector< sp<EffectChain> > chains; 2080 2081 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2082 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2083 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2084 sp<EffectChain> ec = t->mEffectChains[j]; 2085 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2086 chains.push(ec); 2087 } 2088 } 2089 } 2090 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2091 sp<RecordThread> t = mRecordThreads.valueAt(i); 2092 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2093 sp<EffectChain> ec = t->mEffectChains[j]; 2094 chains.push(ec); 2095 } 2096 } 2097 2098 for (size_t i = 0; i < chains.size(); i++) { 2099 sp<EffectChain> ec = chains[i]; 2100 int sessionid = ec->sessionId(); 2101 sp<ThreadBase> t = ec->mThread.promote(); 2102 if (t == 0) { 2103 continue; 2104 } 2105 size_t numsessionrefs = mAudioSessionRefs.size(); 2106 bool found = false; 2107 for (size_t k = 0; k < numsessionrefs; k++) { 2108 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2109 if (ref->mSessionid == sessionid) { 2110 ALOGV(" session %d still exists for %d with %d refs", 2111 sessionid, ref->mPid, ref->mCnt); 2112 found = true; 2113 break; 2114 } 2115 } 2116 if (!found) { 2117 Mutex::Autolock _l(t->mLock); 2118 // remove all effects from the chain 2119 while (ec->mEffects.size()) { 2120 sp<EffectModule> effect = ec->mEffects[0]; 2121 effect->unPin(); 2122 t->removeEffect_l(effect); 2123 if (effect->purgeHandles()) { 2124 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2125 } 2126 AudioSystem::unregisterEffect(effect->id()); 2127 } 2128 } 2129 } 2130 return; 2131} 2132 2133// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2134AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2135{ 2136 return mPlaybackThreads.valueFor(output).get(); 2137} 2138 2139// checkMixerThread_l() must be called with AudioFlinger::mLock held 2140AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2141{ 2142 PlaybackThread *thread = checkPlaybackThread_l(output); 2143 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2144} 2145 2146// checkRecordThread_l() must be called with AudioFlinger::mLock held 2147AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2148{ 2149 return mRecordThreads.valueFor(input).get(); 2150} 2151 2152uint32_t AudioFlinger::nextUniqueId() 2153{ 2154 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2155} 2156 2157AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2158{ 2159 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2160 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2161 AudioStreamOut *output = thread->getOutput(); 2162 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2163 return thread; 2164 } 2165 } 2166 return NULL; 2167} 2168 2169audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2170{ 2171 PlaybackThread *thread = primaryPlaybackThread_l(); 2172 2173 if (thread == NULL) { 2174 return 0; 2175 } 2176 2177 return thread->outDevice(); 2178} 2179 2180sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2181 int triggerSession, 2182 int listenerSession, 2183 sync_event_callback_t callBack, 2184 wp<RefBase> cookie) 2185{ 2186 Mutex::Autolock _l(mLock); 2187 2188 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2189 status_t playStatus = NAME_NOT_FOUND; 2190 status_t recStatus = NAME_NOT_FOUND; 2191 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2192 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2193 if (playStatus == NO_ERROR) { 2194 return event; 2195 } 2196 } 2197 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2198 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2199 if (recStatus == NO_ERROR) { 2200 return event; 2201 } 2202 } 2203 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2204 mPendingSyncEvents.add(event); 2205 } else { 2206 ALOGV("createSyncEvent() invalid event %d", event->type()); 2207 event.clear(); 2208 } 2209 return event; 2210} 2211 2212// ---------------------------------------------------------------------------- 2213// Effect management 2214// ---------------------------------------------------------------------------- 2215 2216 2217status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2218{ 2219 Mutex::Autolock _l(mLock); 2220 return EffectQueryNumberEffects(numEffects); 2221} 2222 2223status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2224{ 2225 Mutex::Autolock _l(mLock); 2226 return EffectQueryEffect(index, descriptor); 2227} 2228 2229status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2230 effect_descriptor_t *descriptor) const 2231{ 2232 Mutex::Autolock _l(mLock); 2233 return EffectGetDescriptor(pUuid, descriptor); 2234} 2235 2236 2237sp<IEffect> AudioFlinger::createEffect( 2238 effect_descriptor_t *pDesc, 2239 const sp<IEffectClient>& effectClient, 2240 int32_t priority, 2241 audio_io_handle_t io, 2242 int sessionId, 2243 status_t *status, 2244 int *id, 2245 int *enabled) 2246{ 2247 status_t lStatus = NO_ERROR; 2248 sp<EffectHandle> handle; 2249 effect_descriptor_t desc; 2250 2251 pid_t pid = IPCThreadState::self()->getCallingPid(); 2252 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2253 pid, effectClient.get(), priority, sessionId, io); 2254 2255 if (pDesc == NULL) { 2256 lStatus = BAD_VALUE; 2257 goto Exit; 2258 } 2259 2260 // check audio settings permission for global effects 2261 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2262 lStatus = PERMISSION_DENIED; 2263 goto Exit; 2264 } 2265 2266 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2267 // that can only be created by audio policy manager (running in same process) 2268 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2269 lStatus = PERMISSION_DENIED; 2270 goto Exit; 2271 } 2272 2273 { 2274 if (!EffectIsNullUuid(&pDesc->uuid)) { 2275 // if uuid is specified, request effect descriptor 2276 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2277 if (lStatus < 0) { 2278 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2279 goto Exit; 2280 } 2281 } else { 2282 // if uuid is not specified, look for an available implementation 2283 // of the required type in effect factory 2284 if (EffectIsNullUuid(&pDesc->type)) { 2285 ALOGW("createEffect() no effect type"); 2286 lStatus = BAD_VALUE; 2287 goto Exit; 2288 } 2289 uint32_t numEffects = 0; 2290 effect_descriptor_t d; 2291 d.flags = 0; // prevent compiler warning 2292 bool found = false; 2293 2294 lStatus = EffectQueryNumberEffects(&numEffects); 2295 if (lStatus < 0) { 2296 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2297 goto Exit; 2298 } 2299 for (uint32_t i = 0; i < numEffects; i++) { 2300 lStatus = EffectQueryEffect(i, &desc); 2301 if (lStatus < 0) { 2302 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2303 continue; 2304 } 2305 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2306 // If matching type found save effect descriptor. If the session is 2307 // 0 and the effect is not auxiliary, continue enumeration in case 2308 // an auxiliary version of this effect type is available 2309 found = true; 2310 d = desc; 2311 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2312 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2313 break; 2314 } 2315 } 2316 } 2317 if (!found) { 2318 lStatus = BAD_VALUE; 2319 ALOGW("createEffect() effect not found"); 2320 goto Exit; 2321 } 2322 // For same effect type, chose auxiliary version over insert version if 2323 // connect to output mix (Compliance to OpenSL ES) 2324 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2325 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2326 desc = d; 2327 } 2328 } 2329 2330 // Do not allow auxiliary effects on a session different from 0 (output mix) 2331 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2332 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2333 lStatus = INVALID_OPERATION; 2334 goto Exit; 2335 } 2336 2337 // check recording permission for visualizer 2338 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2339 !recordingAllowed()) { 2340 lStatus = PERMISSION_DENIED; 2341 goto Exit; 2342 } 2343 2344 // return effect descriptor 2345 *pDesc = desc; 2346 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2347 // if the output returned by getOutputForEffect() is removed before we lock the 2348 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2349 // and we will exit safely 2350 io = AudioSystem::getOutputForEffect(&desc); 2351 ALOGV("createEffect got output %d", io); 2352 } 2353 2354 Mutex::Autolock _l(mLock); 2355 2356 // If output is not specified try to find a matching audio session ID in one of the 2357 // output threads. 2358 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2359 // because of code checking output when entering the function. 2360 // Note: io is never 0 when creating an effect on an input 2361 if (io == AUDIO_IO_HANDLE_NONE) { 2362 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2363 // output must be specified by AudioPolicyManager when using session 2364 // AUDIO_SESSION_OUTPUT_STAGE 2365 lStatus = BAD_VALUE; 2366 goto Exit; 2367 } 2368 // look for the thread where the specified audio session is present 2369 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2370 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2371 io = mPlaybackThreads.keyAt(i); 2372 break; 2373 } 2374 } 2375 if (io == 0) { 2376 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2377 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2378 io = mRecordThreads.keyAt(i); 2379 break; 2380 } 2381 } 2382 } 2383 // If no output thread contains the requested session ID, default to 2384 // first output. The effect chain will be moved to the correct output 2385 // thread when a track with the same session ID is created 2386 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2387 io = mPlaybackThreads.keyAt(0); 2388 } 2389 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2390 } 2391 ThreadBase *thread = checkRecordThread_l(io); 2392 if (thread == NULL) { 2393 thread = checkPlaybackThread_l(io); 2394 if (thread == NULL) { 2395 ALOGE("createEffect() unknown output thread"); 2396 lStatus = BAD_VALUE; 2397 goto Exit; 2398 } 2399 } 2400 2401 sp<Client> client = registerPid(pid); 2402 2403 // create effect on selected output thread 2404 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2405 &desc, enabled, &lStatus); 2406 if (handle != 0 && id != NULL) { 2407 *id = handle->id(); 2408 } 2409 if (handle == 0) { 2410 // remove local strong reference to Client with mClientLock held 2411 Mutex::Autolock _cl(mClientLock); 2412 client.clear(); 2413 } 2414 } 2415 2416Exit: 2417 *status = lStatus; 2418 return handle; 2419} 2420 2421status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2422 audio_io_handle_t dstOutput) 2423{ 2424 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2425 sessionId, srcOutput, dstOutput); 2426 Mutex::Autolock _l(mLock); 2427 if (srcOutput == dstOutput) { 2428 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2429 return NO_ERROR; 2430 } 2431 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2432 if (srcThread == NULL) { 2433 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2434 return BAD_VALUE; 2435 } 2436 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2437 if (dstThread == NULL) { 2438 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2439 return BAD_VALUE; 2440 } 2441 2442 Mutex::Autolock _dl(dstThread->mLock); 2443 Mutex::Autolock _sl(srcThread->mLock); 2444 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2445} 2446 2447// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2448status_t AudioFlinger::moveEffectChain_l(int sessionId, 2449 AudioFlinger::PlaybackThread *srcThread, 2450 AudioFlinger::PlaybackThread *dstThread, 2451 bool reRegister) 2452{ 2453 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2454 sessionId, srcThread, dstThread); 2455 2456 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2457 if (chain == 0) { 2458 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2459 sessionId, srcThread); 2460 return INVALID_OPERATION; 2461 } 2462 2463 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2464 // so that a new chain is created with correct parameters when first effect is added. This is 2465 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2466 // removed. 2467 srcThread->removeEffectChain_l(chain); 2468 2469 // transfer all effects one by one so that new effect chain is created on new thread with 2470 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2471 sp<EffectChain> dstChain; 2472 uint32_t strategy = 0; // prevent compiler warning 2473 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2474 Vector< sp<EffectModule> > removed; 2475 status_t status = NO_ERROR; 2476 while (effect != 0) { 2477 srcThread->removeEffect_l(effect); 2478 removed.add(effect); 2479 status = dstThread->addEffect_l(effect); 2480 if (status != NO_ERROR) { 2481 break; 2482 } 2483 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2484 if (effect->state() == EffectModule::ACTIVE || 2485 effect->state() == EffectModule::STOPPING) { 2486 effect->start(); 2487 } 2488 // if the move request is not received from audio policy manager, the effect must be 2489 // re-registered with the new strategy and output 2490 if (dstChain == 0) { 2491 dstChain = effect->chain().promote(); 2492 if (dstChain == 0) { 2493 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2494 status = NO_INIT; 2495 break; 2496 } 2497 strategy = dstChain->strategy(); 2498 } 2499 if (reRegister) { 2500 AudioSystem::unregisterEffect(effect->id()); 2501 AudioSystem::registerEffect(&effect->desc(), 2502 dstThread->id(), 2503 strategy, 2504 sessionId, 2505 effect->id()); 2506 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2507 } 2508 effect = chain->getEffectFromId_l(0); 2509 } 2510 2511 if (status != NO_ERROR) { 2512 for (size_t i = 0; i < removed.size(); i++) { 2513 srcThread->addEffect_l(removed[i]); 2514 if (dstChain != 0 && reRegister) { 2515 AudioSystem::unregisterEffect(removed[i]->id()); 2516 AudioSystem::registerEffect(&removed[i]->desc(), 2517 srcThread->id(), 2518 strategy, 2519 sessionId, 2520 removed[i]->id()); 2521 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2522 } 2523 } 2524 } 2525 2526 return status; 2527} 2528 2529bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2530{ 2531 if (mGlobalEffectEnableTime != 0 && 2532 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2533 return true; 2534 } 2535 2536 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2537 sp<EffectChain> ec = 2538 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2539 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2540 return true; 2541 } 2542 } 2543 return false; 2544} 2545 2546void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2547{ 2548 Mutex::Autolock _l(mLock); 2549 2550 mGlobalEffectEnableTime = systemTime(); 2551 2552 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2553 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2554 if (t->mType == ThreadBase::OFFLOAD) { 2555 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2556 } 2557 } 2558 2559} 2560 2561struct Entry { 2562#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2563 char mName[MAX_NAME]; 2564}; 2565 2566int comparEntry(const void *p1, const void *p2) 2567{ 2568 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2569} 2570 2571#ifdef TEE_SINK 2572void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2573{ 2574 NBAIO_Source *teeSource = source.get(); 2575 if (teeSource != NULL) { 2576 // .wav rotation 2577 // There is a benign race condition if 2 threads call this simultaneously. 2578 // They would both traverse the directory, but the result would simply be 2579 // failures at unlink() which are ignored. It's also unlikely since 2580 // normally dumpsys is only done by bugreport or from the command line. 2581 char teePath[32+256]; 2582 strcpy(teePath, "/data/misc/media"); 2583 size_t teePathLen = strlen(teePath); 2584 DIR *dir = opendir(teePath); 2585 teePath[teePathLen++] = '/'; 2586 if (dir != NULL) { 2587#define MAX_SORT 20 // number of entries to sort 2588#define MAX_KEEP 10 // number of entries to keep 2589 struct Entry entries[MAX_SORT]; 2590 size_t entryCount = 0; 2591 while (entryCount < MAX_SORT) { 2592 struct dirent de; 2593 struct dirent *result = NULL; 2594 int rc = readdir_r(dir, &de, &result); 2595 if (rc != 0) { 2596 ALOGW("readdir_r failed %d", rc); 2597 break; 2598 } 2599 if (result == NULL) { 2600 break; 2601 } 2602 if (result != &de) { 2603 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2604 break; 2605 } 2606 // ignore non .wav file entries 2607 size_t nameLen = strlen(de.d_name); 2608 if (nameLen <= 4 || nameLen >= MAX_NAME || 2609 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2610 continue; 2611 } 2612 strcpy(entries[entryCount++].mName, de.d_name); 2613 } 2614 (void) closedir(dir); 2615 if (entryCount > MAX_KEEP) { 2616 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2617 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2618 strcpy(&teePath[teePathLen], entries[i].mName); 2619 (void) unlink(teePath); 2620 } 2621 } 2622 } else { 2623 if (fd >= 0) { 2624 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2625 } 2626 } 2627 char teeTime[16]; 2628 struct timeval tv; 2629 gettimeofday(&tv, NULL); 2630 struct tm tm; 2631 localtime_r(&tv.tv_sec, &tm); 2632 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2633 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2634 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2635 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2636 if (teeFd >= 0) { 2637 char wavHeader[44]; 2638 memcpy(wavHeader, 2639 "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", 2640 sizeof(wavHeader)); 2641 NBAIO_Format format = teeSource->format(); 2642 unsigned channelCount = Format_channelCount(format); 2643 ALOG_ASSERT(channelCount <= FCC_2); 2644 uint32_t sampleRate = Format_sampleRate(format); 2645 wavHeader[22] = channelCount; // number of channels 2646 wavHeader[24] = sampleRate; // sample rate 2647 wavHeader[25] = sampleRate >> 8; 2648 wavHeader[32] = channelCount * 2; // block alignment 2649 write(teeFd, wavHeader, sizeof(wavHeader)); 2650 size_t total = 0; 2651 bool firstRead = true; 2652 for (;;) { 2653#define TEE_SINK_READ 1024 2654 short buffer[TEE_SINK_READ * FCC_2]; 2655 size_t count = TEE_SINK_READ; 2656 ssize_t actual = teeSource->read(buffer, count, 2657 AudioBufferProvider::kInvalidPTS); 2658 bool wasFirstRead = firstRead; 2659 firstRead = false; 2660 if (actual <= 0) { 2661 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2662 continue; 2663 } 2664 break; 2665 } 2666 ALOG_ASSERT(actual <= (ssize_t)count); 2667 write(teeFd, buffer, actual * channelCount * sizeof(short)); 2668 total += actual; 2669 } 2670 lseek(teeFd, (off_t) 4, SEEK_SET); 2671 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 2672 write(teeFd, &temp, sizeof(temp)); 2673 lseek(teeFd, (off_t) 40, SEEK_SET); 2674 temp = total * channelCount * sizeof(short); 2675 write(teeFd, &temp, sizeof(temp)); 2676 close(teeFd); 2677 if (fd >= 0) { 2678 dprintf(fd, "tee copied to %s\n", teePath); 2679 } 2680 } else { 2681 if (fd >= 0) { 2682 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2683 } 2684 } 2685 } 2686} 2687#endif 2688 2689// ---------------------------------------------------------------------------- 2690 2691status_t AudioFlinger::onTransact( 2692 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2693{ 2694 return BnAudioFlinger::onTransact(code, data, reply, flags); 2695} 2696 2697}; // namespace android 2698