AudioFlinger.cpp revision e93cc03da360a1a0d2ad937c745ce8c8e8be81c2
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 <memunreachable/memunreachable.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/AudioResamplerPublic.h> 50 51#include <media/EffectsFactoryApi.h> 52#include <audio_effects/effect_visualizer.h> 53#include <audio_effects/effect_ns.h> 54#include <audio_effects/effect_aec.h> 55 56#include <audio_utils/primitives.h> 57 58#include <powermanager/PowerManager.h> 59 60#include <media/IMediaLogService.h> 61 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <mediautils/BatteryNotifier.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (audio_get_main_format(format)) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 case AUDIO_FORMAT_IEC61937: return "iec61937"; 135 default: 136 break; 137 } 138 return "unknown"; 139} 140 141static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 142{ 143 const hw_module_t *mod; 144 int rc; 145 146 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 147 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 148 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 149 if (rc) { 150 goto out; 151 } 152 rc = audio_hw_device_open(mod, dev); 153 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 154 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 155 if (rc) { 156 goto out; 157 } 158 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 159 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 160 rc = BAD_VALUE; 161 goto out; 162 } 163 return 0; 164 165out: 166 *dev = NULL; 167 return rc; 168} 169 170// ---------------------------------------------------------------------------- 171 172AudioFlinger::AudioFlinger() 173 : BnAudioFlinger(), 174 mPrimaryHardwareDev(NULL), 175 mAudioHwDevs(NULL), 176 mHardwareStatus(AUDIO_HW_IDLE), 177 mMasterVolume(1.0f), 178 mMasterMute(false), 179 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 180 mMode(AUDIO_MODE_INVALID), 181 mBtNrecIsOff(false), 182 mIsLowRamDevice(true), 183 mIsDeviceTypeKnown(false), 184 mGlobalEffectEnableTime(0), 185 mSystemReady(false) 186{ 187 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 188 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 189 // zero ID has a special meaning, so unavailable 190 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 191 } 192 193 getpid_cached = getpid(); 194 const bool doLog = property_get_bool("ro.test_harness", false); 195 if (doLog) { 196 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 197 MemoryHeapBase::READ_ONLY); 198 } 199 200 // reset battery stats. 201 // if the audio service has crashed, battery stats could be left 202 // in bad state, reset the state upon service start. 203 BatteryNotifier::getInstance().noteResetAudio(); 204 205#ifdef TEE_SINK 206 char value[PROPERTY_VALUE_MAX]; 207 (void) property_get("ro.debuggable", value, "0"); 208 int debuggable = atoi(value); 209 int teeEnabled = 0; 210 if (debuggable) { 211 (void) property_get("af.tee", value, "0"); 212 teeEnabled = atoi(value); 213 } 214 // FIXME symbolic constants here 215 if (teeEnabled & 1) { 216 mTeeSinkInputEnabled = true; 217 } 218 if (teeEnabled & 2) { 219 mTeeSinkOutputEnabled = true; 220 } 221 if (teeEnabled & 4) { 222 mTeeSinkTrackEnabled = true; 223 } 224#endif 225} 226 227void AudioFlinger::onFirstRef() 228{ 229 Mutex::Autolock _l(mLock); 230 231 /* TODO: move all this work into an Init() function */ 232 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 233 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 234 uint32_t int_val; 235 if (1 == sscanf(val_str, "%u", &int_val)) { 236 mStandbyTimeInNsecs = milliseconds(int_val); 237 ALOGI("Using %u mSec as standby time.", int_val); 238 } else { 239 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 240 ALOGI("Using default %u mSec as standby time.", 241 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 242 } 243 } 244 245 mPatchPanel = new PatchPanel(this); 246 247 mMode = AUDIO_MODE_NORMAL; 248} 249 250AudioFlinger::~AudioFlinger() 251{ 252 while (!mRecordThreads.isEmpty()) { 253 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 254 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 255 } 256 while (!mPlaybackThreads.isEmpty()) { 257 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 258 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 259 } 260 261 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 262 // no mHardwareLock needed, as there are no other references to this 263 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 264 delete mAudioHwDevs.valueAt(i); 265 } 266 267 // Tell media.log service about any old writers that still need to be unregistered 268 if (mLogMemoryDealer != 0) { 269 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 270 if (binder != 0) { 271 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 272 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 273 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 274 mUnregisteredWriters.pop(); 275 mediaLogService->unregisterWriter(iMemory); 276 } 277 } 278 } 279} 280 281static const char * const audio_interfaces[] = { 282 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 283 AUDIO_HARDWARE_MODULE_ID_A2DP, 284 AUDIO_HARDWARE_MODULE_ID_USB, 285}; 286#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 287 288AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 289 audio_module_handle_t module, 290 audio_devices_t devices) 291{ 292 // if module is 0, the request comes from an old policy manager and we should load 293 // well known modules 294 if (module == 0) { 295 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 296 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 297 loadHwModule_l(audio_interfaces[i]); 298 } 299 // then try to find a module supporting the requested device. 300 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 301 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 302 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 303 if ((dev->get_supported_devices != NULL) && 304 (dev->get_supported_devices(dev) & devices) == devices) 305 return audioHwDevice; 306 } 307 } else { 308 // check a match for the requested module handle 309 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 310 if (audioHwDevice != NULL) { 311 return audioHwDevice; 312 } 313 } 314 315 return NULL; 316} 317 318void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 319{ 320 const size_t SIZE = 256; 321 char buffer[SIZE]; 322 String8 result; 323 324 result.append("Clients:\n"); 325 for (size_t i = 0; i < mClients.size(); ++i) { 326 sp<Client> client = mClients.valueAt(i).promote(); 327 if (client != 0) { 328 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 329 result.append(buffer); 330 } 331 } 332 333 result.append("Notification Clients:\n"); 334 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 335 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 336 result.append(buffer); 337 } 338 339 result.append("Global session refs:\n"); 340 result.append(" session pid count\n"); 341 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 342 AudioSessionRef *r = mAudioSessionRefs[i]; 343 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 344 result.append(buffer); 345 } 346 write(fd, result.string(), result.size()); 347} 348 349 350void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 hardware_call_state hardwareStatus = mHardwareStatus; 356 357 snprintf(buffer, SIZE, "Hardware status: %d\n" 358 "Standby Time mSec: %u\n", 359 hardwareStatus, 360 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 361 result.append(buffer); 362 write(fd, result.string(), result.size()); 363} 364 365void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 366{ 367 const size_t SIZE = 256; 368 char buffer[SIZE]; 369 String8 result; 370 snprintf(buffer, SIZE, "Permission Denial: " 371 "can't dump AudioFlinger from pid=%d, uid=%d\n", 372 IPCThreadState::self()->getCallingPid(), 373 IPCThreadState::self()->getCallingUid()); 374 result.append(buffer); 375 write(fd, result.string(), result.size()); 376} 377 378bool AudioFlinger::dumpTryLock(Mutex& mutex) 379{ 380 bool locked = false; 381 for (int i = 0; i < kDumpLockRetries; ++i) { 382 if (mutex.tryLock() == NO_ERROR) { 383 locked = true; 384 break; 385 } 386 usleep(kDumpLockSleepUs); 387 } 388 return locked; 389} 390 391status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 392{ 393 if (!dumpAllowed()) { 394 dumpPermissionDenial(fd, args); 395 } else { 396 // get state of hardware lock 397 bool hardwareLocked = dumpTryLock(mHardwareLock); 398 if (!hardwareLocked) { 399 String8 result(kHardwareLockedString); 400 write(fd, result.string(), result.size()); 401 } else { 402 mHardwareLock.unlock(); 403 } 404 405 bool locked = dumpTryLock(mLock); 406 407 // failed to lock - AudioFlinger is probably deadlocked 408 if (!locked) { 409 String8 result(kDeadlockedString); 410 write(fd, result.string(), result.size()); 411 } 412 413 bool clientLocked = dumpTryLock(mClientLock); 414 if (!clientLocked) { 415 String8 result(kClientLockedString); 416 write(fd, result.string(), result.size()); 417 } 418 419 EffectDumpEffects(fd); 420 421 dumpClients(fd, args); 422 if (clientLocked) { 423 mClientLock.unlock(); 424 } 425 426 dumpInternals(fd, args); 427 428 // dump playback threads 429 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 430 mPlaybackThreads.valueAt(i)->dump(fd, args); 431 } 432 433 // dump record threads 434 for (size_t i = 0; i < mRecordThreads.size(); i++) { 435 mRecordThreads.valueAt(i)->dump(fd, args); 436 } 437 438 // dump orphan effect chains 439 if (mOrphanEffectChains.size() != 0) { 440 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 441 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 442 mOrphanEffectChains.valueAt(i)->dump(fd, args); 443 } 444 } 445 // dump all hardware devs 446 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 447 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 448 dev->dump(dev, fd); 449 } 450 451#ifdef TEE_SINK 452 // dump the serially shared record tee sink 453 if (mRecordTeeSource != 0) { 454 dumpTee(fd, mRecordTeeSource); 455 } 456#endif 457 458 if (locked) { 459 mLock.unlock(); 460 } 461 462 // append a copy of media.log here by forwarding fd to it, but don't attempt 463 // to lookup the service if it's not running, as it will block for a second 464 if (mLogMemoryDealer != 0) { 465 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 466 if (binder != 0) { 467 dprintf(fd, "\nmedia.log:\n"); 468 Vector<String16> args; 469 binder->dump(fd, args); 470 } 471 } 472 473 // check for optional arguments 474 bool unreachableMemory = false; 475 for (const auto &arg : args) { 476 if (arg == String16("--unreachable")) { 477 unreachableMemory = true; 478 } 479 } 480 481 if (unreachableMemory) { 482 dprintf(fd, "\nDumping unreachable memory:\n"); 483 // TODO - should limit be an argument parameter? 484 std::string s = GetUnreachableMemoryString(true /* contents */, 10000 /* limit */); 485 write(fd, s.c_str(), s.size()); 486 } 487 } 488 return NO_ERROR; 489} 490 491sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 492{ 493 Mutex::Autolock _cl(mClientLock); 494 // If pid is already in the mClients wp<> map, then use that entry 495 // (for which promote() is always != 0), otherwise create a new entry and Client. 496 sp<Client> client = mClients.valueFor(pid).promote(); 497 if (client == 0) { 498 client = new Client(this, pid); 499 mClients.add(pid, client); 500 } 501 502 return client; 503} 504 505sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 506{ 507 // If there is no memory allocated for logs, return a dummy writer that does nothing 508 if (mLogMemoryDealer == 0) { 509 return new NBLog::Writer(); 510 } 511 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 512 // Similarly if we can't contact the media.log service, also return a dummy writer 513 if (binder == 0) { 514 return new NBLog::Writer(); 515 } 516 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 517 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 518 // If allocation fails, consult the vector of previously unregistered writers 519 // and garbage-collect one or more them until an allocation succeeds 520 if (shared == 0) { 521 Mutex::Autolock _l(mUnregisteredWritersLock); 522 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 523 { 524 // Pick the oldest stale writer to garbage-collect 525 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 526 mUnregisteredWriters.removeAt(0); 527 mediaLogService->unregisterWriter(iMemory); 528 // Now the media.log remote reference to IMemory is gone. When our last local 529 // reference to IMemory also drops to zero at end of this block, 530 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 531 } 532 // Re-attempt the allocation 533 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 534 if (shared != 0) { 535 goto success; 536 } 537 } 538 // Even after garbage-collecting all old writers, there is still not enough memory, 539 // so return a dummy writer 540 return new NBLog::Writer(); 541 } 542success: 543 mediaLogService->registerWriter(shared, size, name); 544 return new NBLog::Writer(size, shared); 545} 546 547void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 548{ 549 if (writer == 0) { 550 return; 551 } 552 sp<IMemory> iMemory(writer->getIMemory()); 553 if (iMemory == 0) { 554 return; 555 } 556 // Rather than removing the writer immediately, append it to a queue of old writers to 557 // be garbage-collected later. This allows us to continue to view old logs for a while. 558 Mutex::Autolock _l(mUnregisteredWritersLock); 559 mUnregisteredWriters.push(writer); 560} 561 562// IAudioFlinger interface 563 564 565sp<IAudioTrack> AudioFlinger::createTrack( 566 audio_stream_type_t streamType, 567 uint32_t sampleRate, 568 audio_format_t format, 569 audio_channel_mask_t channelMask, 570 size_t *frameCount, 571 IAudioFlinger::track_flags_t *flags, 572 const sp<IMemory>& sharedBuffer, 573 audio_io_handle_t output, 574 pid_t tid, 575 audio_session_t *sessionId, 576 int clientUid, 577 status_t *status) 578{ 579 sp<PlaybackThread::Track> track; 580 sp<TrackHandle> trackHandle; 581 sp<Client> client; 582 status_t lStatus; 583 audio_session_t lSessionId; 584 585 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 586 // but if someone uses binder directly they could bypass that and cause us to crash 587 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 588 ALOGE("createTrack() invalid stream type %d", streamType); 589 lStatus = BAD_VALUE; 590 goto Exit; 591 } 592 593 // further sample rate checks are performed by createTrack_l() depending on the thread type 594 if (sampleRate == 0) { 595 ALOGE("createTrack() invalid sample rate %u", sampleRate); 596 lStatus = BAD_VALUE; 597 goto Exit; 598 } 599 600 // further channel mask checks are performed by createTrack_l() depending on the thread type 601 if (!audio_is_output_channel(channelMask)) { 602 ALOGE("createTrack() invalid channel mask %#x", channelMask); 603 lStatus = BAD_VALUE; 604 goto Exit; 605 } 606 607 // further format checks are performed by createTrack_l() depending on the thread type 608 if (!audio_is_valid_format(format)) { 609 ALOGE("createTrack() invalid format %#x", format); 610 lStatus = BAD_VALUE; 611 goto Exit; 612 } 613 614 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 615 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 616 lStatus = BAD_VALUE; 617 goto Exit; 618 } 619 620 { 621 Mutex::Autolock _l(mLock); 622 PlaybackThread *thread = checkPlaybackThread_l(output); 623 if (thread == NULL) { 624 ALOGE("no playback thread found for output handle %d", output); 625 lStatus = BAD_VALUE; 626 goto Exit; 627 } 628 629 pid_t pid = IPCThreadState::self()->getCallingPid(); 630 client = registerPid(pid); 631 632 PlaybackThread *effectThread = NULL; 633 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 634 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 635 ALOGE("createTrack() invalid session ID %d", *sessionId); 636 lStatus = BAD_VALUE; 637 goto Exit; 638 } 639 lSessionId = *sessionId; 640 // check if an effect chain with the same session ID is present on another 641 // output thread and move it here. 642 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 643 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 644 if (mPlaybackThreads.keyAt(i) != output) { 645 uint32_t sessions = t->hasAudioSession(lSessionId); 646 if (sessions & PlaybackThread::EFFECT_SESSION) { 647 effectThread = t.get(); 648 break; 649 } 650 } 651 } 652 } else { 653 // if no audio session id is provided, create one here 654 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 655 if (sessionId != NULL) { 656 *sessionId = lSessionId; 657 } 658 } 659 ALOGV("createTrack() lSessionId: %d", lSessionId); 660 661 track = thread->createTrack_l(client, streamType, sampleRate, format, 662 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 663 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 664 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 665 666 // move effect chain to this output thread if an effect on same session was waiting 667 // for a track to be created 668 if (lStatus == NO_ERROR && effectThread != NULL) { 669 // no risk of deadlock because AudioFlinger::mLock is held 670 Mutex::Autolock _dl(thread->mLock); 671 Mutex::Autolock _sl(effectThread->mLock); 672 moveEffectChain_l(lSessionId, effectThread, thread, true); 673 } 674 675 // Look for sync events awaiting for a session to be used. 676 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 677 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 678 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 679 if (lStatus == NO_ERROR) { 680 (void) track->setSyncEvent(mPendingSyncEvents[i]); 681 } else { 682 mPendingSyncEvents[i]->cancel(); 683 } 684 mPendingSyncEvents.removeAt(i); 685 i--; 686 } 687 } 688 } 689 690 setAudioHwSyncForSession_l(thread, lSessionId); 691 } 692 693 if (lStatus != NO_ERROR) { 694 // remove local strong reference to Client before deleting the Track so that the 695 // Client destructor is called by the TrackBase destructor with mClientLock held 696 // Don't hold mClientLock when releasing the reference on the track as the 697 // destructor will acquire it. 698 { 699 Mutex::Autolock _cl(mClientLock); 700 client.clear(); 701 } 702 track.clear(); 703 goto Exit; 704 } 705 706 // return handle to client 707 trackHandle = new TrackHandle(track); 708 709Exit: 710 *status = lStatus; 711 return trackHandle; 712} 713 714uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 715{ 716 Mutex::Autolock _l(mLock); 717 ThreadBase *thread = checkThread_l(ioHandle); 718 if (thread == NULL) { 719 ALOGW("sampleRate() unknown thread %d", ioHandle); 720 return 0; 721 } 722 return thread->sampleRate(); 723} 724 725audio_format_t AudioFlinger::format(audio_io_handle_t output) const 726{ 727 Mutex::Autolock _l(mLock); 728 PlaybackThread *thread = checkPlaybackThread_l(output); 729 if (thread == NULL) { 730 ALOGW("format() unknown thread %d", output); 731 return AUDIO_FORMAT_INVALID; 732 } 733 return thread->format(); 734} 735 736size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 737{ 738 Mutex::Autolock _l(mLock); 739 ThreadBase *thread = checkThread_l(ioHandle); 740 if (thread == NULL) { 741 ALOGW("frameCount() unknown thread %d", ioHandle); 742 return 0; 743 } 744 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 745 // should examine all callers and fix them to handle smaller counts 746 return thread->frameCount(); 747} 748 749size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 750{ 751 Mutex::Autolock _l(mLock); 752 ThreadBase *thread = checkThread_l(ioHandle); 753 if (thread == NULL) { 754 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 755 return 0; 756 } 757 return thread->frameCountHAL(); 758} 759 760uint32_t AudioFlinger::latency(audio_io_handle_t output) const 761{ 762 Mutex::Autolock _l(mLock); 763 PlaybackThread *thread = checkPlaybackThread_l(output); 764 if (thread == NULL) { 765 ALOGW("latency(): no playback thread found for output handle %d", output); 766 return 0; 767 } 768 return thread->latency(); 769} 770 771status_t AudioFlinger::setMasterVolume(float value) 772{ 773 status_t ret = initCheck(); 774 if (ret != NO_ERROR) { 775 return ret; 776 } 777 778 // check calling permissions 779 if (!settingsAllowed()) { 780 return PERMISSION_DENIED; 781 } 782 783 Mutex::Autolock _l(mLock); 784 mMasterVolume = value; 785 786 // Set master volume in the HALs which support it. 787 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 788 AutoMutex lock(mHardwareLock); 789 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 790 791 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 792 if (dev->canSetMasterVolume()) { 793 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 794 } 795 mHardwareStatus = AUDIO_HW_IDLE; 796 } 797 798 // Now set the master volume in each playback thread. Playback threads 799 // assigned to HALs which do not have master volume support will apply 800 // master volume during the mix operation. Threads with HALs which do 801 // support master volume will simply ignore the setting. 802 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 803 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 804 continue; 805 } 806 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 807 } 808 809 return NO_ERROR; 810} 811 812status_t AudioFlinger::setMode(audio_mode_t mode) 813{ 814 status_t ret = initCheck(); 815 if (ret != NO_ERROR) { 816 return ret; 817 } 818 819 // check calling permissions 820 if (!settingsAllowed()) { 821 return PERMISSION_DENIED; 822 } 823 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 824 ALOGW("Illegal value: setMode(%d)", mode); 825 return BAD_VALUE; 826 } 827 828 { // scope for the lock 829 AutoMutex lock(mHardwareLock); 830 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 831 mHardwareStatus = AUDIO_HW_SET_MODE; 832 ret = dev->set_mode(dev, mode); 833 mHardwareStatus = AUDIO_HW_IDLE; 834 } 835 836 if (NO_ERROR == ret) { 837 Mutex::Autolock _l(mLock); 838 mMode = mode; 839 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 840 mPlaybackThreads.valueAt(i)->setMode(mode); 841 } 842 843 return ret; 844} 845 846status_t AudioFlinger::setMicMute(bool state) 847{ 848 status_t ret = initCheck(); 849 if (ret != NO_ERROR) { 850 return ret; 851 } 852 853 // check calling permissions 854 if (!settingsAllowed()) { 855 return PERMISSION_DENIED; 856 } 857 858 AutoMutex lock(mHardwareLock); 859 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 860 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 861 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 862 status_t result = dev->set_mic_mute(dev, state); 863 if (result != NO_ERROR) { 864 ret = result; 865 } 866 } 867 mHardwareStatus = AUDIO_HW_IDLE; 868 return ret; 869} 870 871bool AudioFlinger::getMicMute() const 872{ 873 status_t ret = initCheck(); 874 if (ret != NO_ERROR) { 875 return false; 876 } 877 bool mute = true; 878 bool state = AUDIO_MODE_INVALID; 879 AutoMutex lock(mHardwareLock); 880 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 881 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 882 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 883 status_t result = dev->get_mic_mute(dev, &state); 884 if (result == NO_ERROR) { 885 mute = mute && state; 886 } 887 } 888 mHardwareStatus = AUDIO_HW_IDLE; 889 890 return mute; 891} 892 893status_t AudioFlinger::setMasterMute(bool muted) 894{ 895 status_t ret = initCheck(); 896 if (ret != NO_ERROR) { 897 return ret; 898 } 899 900 // check calling permissions 901 if (!settingsAllowed()) { 902 return PERMISSION_DENIED; 903 } 904 905 Mutex::Autolock _l(mLock); 906 mMasterMute = muted; 907 908 // Set master mute in the HALs which support it. 909 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 910 AutoMutex lock(mHardwareLock); 911 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 912 913 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 914 if (dev->canSetMasterMute()) { 915 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 916 } 917 mHardwareStatus = AUDIO_HW_IDLE; 918 } 919 920 // Now set the master mute in each playback thread. Playback threads 921 // assigned to HALs which do not have master mute support will apply master 922 // mute during the mix operation. Threads with HALs which do support master 923 // mute will simply ignore the setting. 924 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 925 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 926 continue; 927 } 928 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 929 } 930 931 return NO_ERROR; 932} 933 934float AudioFlinger::masterVolume() const 935{ 936 Mutex::Autolock _l(mLock); 937 return masterVolume_l(); 938} 939 940bool AudioFlinger::masterMute() const 941{ 942 Mutex::Autolock _l(mLock); 943 return masterMute_l(); 944} 945 946float AudioFlinger::masterVolume_l() const 947{ 948 return mMasterVolume; 949} 950 951bool AudioFlinger::masterMute_l() const 952{ 953 return mMasterMute; 954} 955 956status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 957{ 958 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 959 ALOGW("setStreamVolume() invalid stream %d", stream); 960 return BAD_VALUE; 961 } 962 pid_t caller = IPCThreadState::self()->getCallingPid(); 963 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 964 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 965 return PERMISSION_DENIED; 966 } 967 968 return NO_ERROR; 969} 970 971status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 972 audio_io_handle_t output) 973{ 974 // check calling permissions 975 if (!settingsAllowed()) { 976 return PERMISSION_DENIED; 977 } 978 979 status_t status = checkStreamType(stream); 980 if (status != NO_ERROR) { 981 return status; 982 } 983 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 984 985 AutoMutex lock(mLock); 986 PlaybackThread *thread = NULL; 987 if (output != AUDIO_IO_HANDLE_NONE) { 988 thread = checkPlaybackThread_l(output); 989 if (thread == NULL) { 990 return BAD_VALUE; 991 } 992 } 993 994 mStreamTypes[stream].volume = value; 995 996 if (thread == NULL) { 997 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 998 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 999 } 1000 } else { 1001 thread->setStreamVolume(stream, value); 1002 } 1003 1004 return NO_ERROR; 1005} 1006 1007status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1008{ 1009 // check calling permissions 1010 if (!settingsAllowed()) { 1011 return PERMISSION_DENIED; 1012 } 1013 1014 status_t status = checkStreamType(stream); 1015 if (status != NO_ERROR) { 1016 return status; 1017 } 1018 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1019 1020 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1021 ALOGE("setStreamMute() invalid stream %d", stream); 1022 return BAD_VALUE; 1023 } 1024 1025 AutoMutex lock(mLock); 1026 mStreamTypes[stream].mute = muted; 1027 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1028 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1029 1030 return NO_ERROR; 1031} 1032 1033float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1034{ 1035 status_t status = checkStreamType(stream); 1036 if (status != NO_ERROR) { 1037 return 0.0f; 1038 } 1039 1040 AutoMutex lock(mLock); 1041 float volume; 1042 if (output != AUDIO_IO_HANDLE_NONE) { 1043 PlaybackThread *thread = checkPlaybackThread_l(output); 1044 if (thread == NULL) { 1045 return 0.0f; 1046 } 1047 volume = thread->streamVolume(stream); 1048 } else { 1049 volume = streamVolume_l(stream); 1050 } 1051 1052 return volume; 1053} 1054 1055bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1056{ 1057 status_t status = checkStreamType(stream); 1058 if (status != NO_ERROR) { 1059 return true; 1060 } 1061 1062 AutoMutex lock(mLock); 1063 return streamMute_l(stream); 1064} 1065 1066 1067void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1068{ 1069 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1070 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1071 } 1072} 1073 1074status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1075{ 1076 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1077 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1078 1079 // check calling permissions 1080 if (!settingsAllowed()) { 1081 return PERMISSION_DENIED; 1082 } 1083 1084 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1085 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1086 Mutex::Autolock _l(mLock); 1087 status_t final_result = NO_ERROR; 1088 { 1089 AutoMutex lock(mHardwareLock); 1090 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1091 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1092 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1093 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1094 final_result = result ?: final_result; 1095 } 1096 mHardwareStatus = AUDIO_HW_IDLE; 1097 } 1098 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1099 AudioParameter param = AudioParameter(keyValuePairs); 1100 String8 value; 1101 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1102 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1103 if (mBtNrecIsOff != btNrecIsOff) { 1104 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1105 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1106 audio_devices_t device = thread->inDevice(); 1107 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1108 // collect all of the thread's session IDs 1109 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1110 // suspend effects associated with those session IDs 1111 for (size_t j = 0; j < ids.size(); ++j) { 1112 audio_session_t sessionId = ids.keyAt(j); 1113 thread->setEffectSuspended(FX_IID_AEC, 1114 suspend, 1115 sessionId); 1116 thread->setEffectSuspended(FX_IID_NS, 1117 suspend, 1118 sessionId); 1119 } 1120 } 1121 mBtNrecIsOff = btNrecIsOff; 1122 } 1123 } 1124 String8 screenState; 1125 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1126 bool isOff = screenState == "off"; 1127 if (isOff != (AudioFlinger::mScreenState & 1)) { 1128 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1129 } 1130 } 1131 return final_result; 1132 } 1133 1134 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1135 // and the thread is exited once the lock is released 1136 sp<ThreadBase> thread; 1137 { 1138 Mutex::Autolock _l(mLock); 1139 thread = checkPlaybackThread_l(ioHandle); 1140 if (thread == 0) { 1141 thread = checkRecordThread_l(ioHandle); 1142 } else if (thread == primaryPlaybackThread_l()) { 1143 // indicate output device change to all input threads for pre processing 1144 AudioParameter param = AudioParameter(keyValuePairs); 1145 int value; 1146 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1147 (value != 0)) { 1148 broacastParametersToRecordThreads_l(keyValuePairs); 1149 } 1150 } 1151 } 1152 if (thread != 0) { 1153 return thread->setParameters(keyValuePairs); 1154 } 1155 return BAD_VALUE; 1156} 1157 1158String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1159{ 1160 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1161 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1162 1163 Mutex::Autolock _l(mLock); 1164 1165 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1166 String8 out_s8; 1167 1168 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1169 char *s; 1170 { 1171 AutoMutex lock(mHardwareLock); 1172 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1173 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1174 s = dev->get_parameters(dev, keys.string()); 1175 mHardwareStatus = AUDIO_HW_IDLE; 1176 } 1177 out_s8 += String8(s ? s : ""); 1178 free(s); 1179 } 1180 return out_s8; 1181 } 1182 1183 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1184 if (playbackThread != NULL) { 1185 return playbackThread->getParameters(keys); 1186 } 1187 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1188 if (recordThread != NULL) { 1189 return recordThread->getParameters(keys); 1190 } 1191 return String8(""); 1192} 1193 1194size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1195 audio_channel_mask_t channelMask) const 1196{ 1197 status_t ret = initCheck(); 1198 if (ret != NO_ERROR) { 1199 return 0; 1200 } 1201 if ((sampleRate == 0) || 1202 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1203 !audio_is_input_channel(channelMask)) { 1204 return 0; 1205 } 1206 1207 AutoMutex lock(mHardwareLock); 1208 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1209 audio_config_t config, proposed; 1210 memset(&proposed, 0, sizeof(proposed)); 1211 proposed.sample_rate = sampleRate; 1212 proposed.channel_mask = channelMask; 1213 proposed.format = format; 1214 1215 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1216 size_t frames; 1217 for (;;) { 1218 // Note: config is currently a const parameter for get_input_buffer_size() 1219 // but we use a copy from proposed in case config changes from the call. 1220 config = proposed; 1221 frames = dev->get_input_buffer_size(dev, &config); 1222 if (frames != 0) { 1223 break; // hal success, config is the result 1224 } 1225 // change one parameter of the configuration each iteration to a more "common" value 1226 // to see if the device will support it. 1227 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1228 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1229 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1230 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1231 } else { 1232 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1233 "format %#x, channelMask 0x%X", 1234 sampleRate, format, channelMask); 1235 break; // retries failed, break out of loop with frames == 0. 1236 } 1237 } 1238 mHardwareStatus = AUDIO_HW_IDLE; 1239 if (frames > 0 && config.sample_rate != sampleRate) { 1240 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1241 } 1242 return frames; // may be converted to bytes at the Java level. 1243} 1244 1245uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1246{ 1247 Mutex::Autolock _l(mLock); 1248 1249 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1250 if (recordThread != NULL) { 1251 return recordThread->getInputFramesLost(); 1252 } 1253 return 0; 1254} 1255 1256status_t AudioFlinger::setVoiceVolume(float value) 1257{ 1258 status_t ret = initCheck(); 1259 if (ret != NO_ERROR) { 1260 return ret; 1261 } 1262 1263 // check calling permissions 1264 if (!settingsAllowed()) { 1265 return PERMISSION_DENIED; 1266 } 1267 1268 AutoMutex lock(mHardwareLock); 1269 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1270 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1271 ret = dev->set_voice_volume(dev, value); 1272 mHardwareStatus = AUDIO_HW_IDLE; 1273 1274 return ret; 1275} 1276 1277status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1278 audio_io_handle_t output) const 1279{ 1280 Mutex::Autolock _l(mLock); 1281 1282 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1283 if (playbackThread != NULL) { 1284 return playbackThread->getRenderPosition(halFrames, dspFrames); 1285 } 1286 1287 return BAD_VALUE; 1288} 1289 1290void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1291{ 1292 Mutex::Autolock _l(mLock); 1293 if (client == 0) { 1294 return; 1295 } 1296 pid_t pid = IPCThreadState::self()->getCallingPid(); 1297 { 1298 Mutex::Autolock _cl(mClientLock); 1299 if (mNotificationClients.indexOfKey(pid) < 0) { 1300 sp<NotificationClient> notificationClient = new NotificationClient(this, 1301 client, 1302 pid); 1303 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1304 1305 mNotificationClients.add(pid, notificationClient); 1306 1307 sp<IBinder> binder = IInterface::asBinder(client); 1308 binder->linkToDeath(notificationClient); 1309 } 1310 } 1311 1312 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1313 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1314 // the config change is always sent from playback or record threads to avoid deadlock 1315 // with AudioSystem::gLock 1316 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1317 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1318 } 1319 1320 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1321 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1322 } 1323} 1324 1325void AudioFlinger::removeNotificationClient(pid_t pid) 1326{ 1327 Mutex::Autolock _l(mLock); 1328 { 1329 Mutex::Autolock _cl(mClientLock); 1330 mNotificationClients.removeItem(pid); 1331 } 1332 1333 ALOGV("%d died, releasing its sessions", pid); 1334 size_t num = mAudioSessionRefs.size(); 1335 bool removed = false; 1336 for (size_t i = 0; i< num; ) { 1337 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1338 ALOGV(" pid %d @ %zu", ref->mPid, i); 1339 if (ref->mPid == pid) { 1340 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1341 mAudioSessionRefs.removeAt(i); 1342 delete ref; 1343 removed = true; 1344 num--; 1345 } else { 1346 i++; 1347 } 1348 } 1349 if (removed) { 1350 purgeStaleEffects_l(); 1351 } 1352} 1353 1354void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1355 const sp<AudioIoDescriptor>& ioDesc, 1356 pid_t pid) 1357{ 1358 Mutex::Autolock _l(mClientLock); 1359 size_t size = mNotificationClients.size(); 1360 for (size_t i = 0; i < size; i++) { 1361 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1362 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1363 } 1364 } 1365} 1366 1367// removeClient_l() must be called with AudioFlinger::mClientLock held 1368void AudioFlinger::removeClient_l(pid_t pid) 1369{ 1370 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1371 IPCThreadState::self()->getCallingPid()); 1372 mClients.removeItem(pid); 1373} 1374 1375// getEffectThread_l() must be called with AudioFlinger::mLock held 1376sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1377 int EffectId) 1378{ 1379 sp<PlaybackThread> thread; 1380 1381 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1382 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1383 ALOG_ASSERT(thread == 0); 1384 thread = mPlaybackThreads.valueAt(i); 1385 } 1386 } 1387 1388 return thread; 1389} 1390 1391 1392 1393// ---------------------------------------------------------------------------- 1394 1395AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1396 : RefBase(), 1397 mAudioFlinger(audioFlinger), 1398 mPid(pid) 1399{ 1400 size_t heapSize = kClientSharedHeapSizeBytes; 1401 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1402 // invalidated tracks 1403 if (!audioFlinger->isLowRamDevice()) { 1404 heapSize *= kClientSharedHeapSizeMultiplier; 1405 } 1406 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1407} 1408 1409// Client destructor must be called with AudioFlinger::mClientLock held 1410AudioFlinger::Client::~Client() 1411{ 1412 mAudioFlinger->removeClient_l(mPid); 1413} 1414 1415sp<MemoryDealer> AudioFlinger::Client::heap() const 1416{ 1417 return mMemoryDealer; 1418} 1419 1420// ---------------------------------------------------------------------------- 1421 1422AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1423 const sp<IAudioFlingerClient>& client, 1424 pid_t pid) 1425 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1426{ 1427} 1428 1429AudioFlinger::NotificationClient::~NotificationClient() 1430{ 1431} 1432 1433void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1434{ 1435 sp<NotificationClient> keep(this); 1436 mAudioFlinger->removeNotificationClient(mPid); 1437} 1438 1439 1440// ---------------------------------------------------------------------------- 1441 1442sp<IAudioRecord> AudioFlinger::openRecord( 1443 audio_io_handle_t input, 1444 uint32_t sampleRate, 1445 audio_format_t format, 1446 audio_channel_mask_t channelMask, 1447 const String16& opPackageName, 1448 size_t *frameCount, 1449 IAudioFlinger::track_flags_t *flags, 1450 pid_t tid, 1451 int clientUid, 1452 audio_session_t *sessionId, 1453 size_t *notificationFrames, 1454 sp<IMemory>& cblk, 1455 sp<IMemory>& buffers, 1456 status_t *status) 1457{ 1458 sp<RecordThread::RecordTrack> recordTrack; 1459 sp<RecordHandle> recordHandle; 1460 sp<Client> client; 1461 status_t lStatus; 1462 audio_session_t lSessionId; 1463 1464 cblk.clear(); 1465 buffers.clear(); 1466 1467 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1468 if (!isTrustedCallingUid(callingUid)) { 1469 ALOGW_IF((uid_t)clientUid != callingUid, 1470 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1471 clientUid = callingUid; 1472 } 1473 1474 // check calling permissions 1475 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1476 ALOGE("openRecord() permission denied: recording not allowed"); 1477 lStatus = PERMISSION_DENIED; 1478 goto Exit; 1479 } 1480 1481 // further sample rate checks are performed by createRecordTrack_l() 1482 if (sampleRate == 0) { 1483 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1484 lStatus = BAD_VALUE; 1485 goto Exit; 1486 } 1487 1488 // we don't yet support anything other than linear PCM 1489 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1490 ALOGE("openRecord() invalid format %#x", format); 1491 lStatus = BAD_VALUE; 1492 goto Exit; 1493 } 1494 1495 // further channel mask checks are performed by createRecordTrack_l() 1496 if (!audio_is_input_channel(channelMask)) { 1497 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1498 lStatus = BAD_VALUE; 1499 goto Exit; 1500 } 1501 1502 { 1503 Mutex::Autolock _l(mLock); 1504 RecordThread *thread = checkRecordThread_l(input); 1505 if (thread == NULL) { 1506 ALOGE("openRecord() checkRecordThread_l failed"); 1507 lStatus = BAD_VALUE; 1508 goto Exit; 1509 } 1510 1511 pid_t pid = IPCThreadState::self()->getCallingPid(); 1512 client = registerPid(pid); 1513 1514 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1515 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1516 lStatus = BAD_VALUE; 1517 goto Exit; 1518 } 1519 lSessionId = *sessionId; 1520 } else { 1521 // if no audio session id is provided, create one here 1522 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1523 if (sessionId != NULL) { 1524 *sessionId = lSessionId; 1525 } 1526 } 1527 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1528 1529 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1530 frameCount, lSessionId, notificationFrames, 1531 clientUid, flags, tid, &lStatus); 1532 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1533 1534 if (lStatus == NO_ERROR) { 1535 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1536 // session and move it to this thread. 1537 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1538 if (chain != 0) { 1539 Mutex::Autolock _l(thread->mLock); 1540 thread->addEffectChain_l(chain); 1541 } 1542 } 1543 } 1544 1545 if (lStatus != NO_ERROR) { 1546 // remove local strong reference to Client before deleting the RecordTrack so that the 1547 // Client destructor is called by the TrackBase destructor with mClientLock held 1548 // Don't hold mClientLock when releasing the reference on the track as the 1549 // destructor will acquire it. 1550 { 1551 Mutex::Autolock _cl(mClientLock); 1552 client.clear(); 1553 } 1554 recordTrack.clear(); 1555 goto Exit; 1556 } 1557 1558 cblk = recordTrack->getCblk(); 1559 buffers = recordTrack->getBuffers(); 1560 1561 // return handle to client 1562 recordHandle = new RecordHandle(recordTrack); 1563 1564Exit: 1565 *status = lStatus; 1566 return recordHandle; 1567} 1568 1569 1570 1571// ---------------------------------------------------------------------------- 1572 1573audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1574{ 1575 if (name == NULL) { 1576 return AUDIO_MODULE_HANDLE_NONE; 1577 } 1578 if (!settingsAllowed()) { 1579 return AUDIO_MODULE_HANDLE_NONE; 1580 } 1581 Mutex::Autolock _l(mLock); 1582 return loadHwModule_l(name); 1583} 1584 1585// loadHwModule_l() must be called with AudioFlinger::mLock held 1586audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1587{ 1588 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1589 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1590 ALOGW("loadHwModule() module %s already loaded", name); 1591 return mAudioHwDevs.keyAt(i); 1592 } 1593 } 1594 1595 audio_hw_device_t *dev; 1596 1597 int rc = load_audio_interface(name, &dev); 1598 if (rc) { 1599 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1600 return AUDIO_MODULE_HANDLE_NONE; 1601 } 1602 1603 mHardwareStatus = AUDIO_HW_INIT; 1604 rc = dev->init_check(dev); 1605 mHardwareStatus = AUDIO_HW_IDLE; 1606 if (rc) { 1607 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1608 return AUDIO_MODULE_HANDLE_NONE; 1609 } 1610 1611 // Check and cache this HAL's level of support for master mute and master 1612 // volume. If this is the first HAL opened, and it supports the get 1613 // methods, use the initial values provided by the HAL as the current 1614 // master mute and volume settings. 1615 1616 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1617 { // scope for auto-lock pattern 1618 AutoMutex lock(mHardwareLock); 1619 1620 if (0 == mAudioHwDevs.size()) { 1621 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1622 if (NULL != dev->get_master_volume) { 1623 float mv; 1624 if (OK == dev->get_master_volume(dev, &mv)) { 1625 mMasterVolume = mv; 1626 } 1627 } 1628 1629 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1630 if (NULL != dev->get_master_mute) { 1631 bool mm; 1632 if (OK == dev->get_master_mute(dev, &mm)) { 1633 mMasterMute = mm; 1634 } 1635 } 1636 } 1637 1638 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1639 if ((NULL != dev->set_master_volume) && 1640 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1641 flags = static_cast<AudioHwDevice::Flags>(flags | 1642 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1643 } 1644 1645 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1646 if ((NULL != dev->set_master_mute) && 1647 (OK == dev->set_master_mute(dev, mMasterMute))) { 1648 flags = static_cast<AudioHwDevice::Flags>(flags | 1649 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1650 } 1651 1652 mHardwareStatus = AUDIO_HW_IDLE; 1653 } 1654 1655 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1656 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1657 1658 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1659 name, dev->common.module->name, dev->common.module->id, handle); 1660 1661 return handle; 1662 1663} 1664 1665// ---------------------------------------------------------------------------- 1666 1667uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1668{ 1669 Mutex::Autolock _l(mLock); 1670 PlaybackThread *thread = primaryPlaybackThread_l(); 1671 return thread != NULL ? thread->sampleRate() : 0; 1672} 1673 1674size_t AudioFlinger::getPrimaryOutputFrameCount() 1675{ 1676 Mutex::Autolock _l(mLock); 1677 PlaybackThread *thread = primaryPlaybackThread_l(); 1678 return thread != NULL ? thread->frameCountHAL() : 0; 1679} 1680 1681// ---------------------------------------------------------------------------- 1682 1683status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1684{ 1685 uid_t uid = IPCThreadState::self()->getCallingUid(); 1686 if (uid != AID_SYSTEM) { 1687 return PERMISSION_DENIED; 1688 } 1689 Mutex::Autolock _l(mLock); 1690 if (mIsDeviceTypeKnown) { 1691 return INVALID_OPERATION; 1692 } 1693 mIsLowRamDevice = isLowRamDevice; 1694 mIsDeviceTypeKnown = true; 1695 return NO_ERROR; 1696} 1697 1698audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 1702 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1703 if (index >= 0) { 1704 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1705 mHwAvSyncIds.valueAt(index), sessionId); 1706 return mHwAvSyncIds.valueAt(index); 1707 } 1708 1709 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1710 if (dev == NULL) { 1711 return AUDIO_HW_SYNC_INVALID; 1712 } 1713 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1714 AudioParameter param = AudioParameter(String8(reply)); 1715 free(reply); 1716 1717 int value; 1718 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1719 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1720 return AUDIO_HW_SYNC_INVALID; 1721 } 1722 1723 // allow only one session for a given HW A/V sync ID. 1724 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1725 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1726 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1727 value, mHwAvSyncIds.keyAt(i)); 1728 mHwAvSyncIds.removeItemsAt(i); 1729 break; 1730 } 1731 } 1732 1733 mHwAvSyncIds.add(sessionId, value); 1734 1735 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1736 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1737 uint32_t sessions = thread->hasAudioSession(sessionId); 1738 if (sessions & PlaybackThread::TRACK_SESSION) { 1739 AudioParameter param = AudioParameter(); 1740 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1741 thread->setParameters(param.toString()); 1742 break; 1743 } 1744 } 1745 1746 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1747 return (audio_hw_sync_t)value; 1748} 1749 1750status_t AudioFlinger::systemReady() 1751{ 1752 Mutex::Autolock _l(mLock); 1753 ALOGI("%s", __FUNCTION__); 1754 if (mSystemReady) { 1755 ALOGW("%s called twice", __FUNCTION__); 1756 return NO_ERROR; 1757 } 1758 mSystemReady = true; 1759 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1760 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1761 thread->systemReady(); 1762 } 1763 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1764 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1765 thread->systemReady(); 1766 } 1767 return NO_ERROR; 1768} 1769 1770// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1771void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1772{ 1773 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1774 if (index >= 0) { 1775 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1776 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1777 AudioParameter param = AudioParameter(); 1778 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1779 thread->setParameters(param.toString()); 1780 } 1781} 1782 1783 1784// ---------------------------------------------------------------------------- 1785 1786 1787sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1788 audio_io_handle_t *output, 1789 audio_config_t *config, 1790 audio_devices_t devices, 1791 const String8& address, 1792 audio_output_flags_t flags) 1793{ 1794 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1795 if (outHwDev == NULL) { 1796 return 0; 1797 } 1798 1799 if (*output == AUDIO_IO_HANDLE_NONE) { 1800 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1801 } else { 1802 // Audio Policy does not currently request a specific output handle. 1803 // If this is ever needed, see openInput_l() for example code. 1804 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1805 return 0; 1806 } 1807 1808 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1809 1810 // FOR TESTING ONLY: 1811 // This if statement allows overriding the audio policy settings 1812 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1813 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1814 // Check only for Normal Mixing mode 1815 if (kEnableExtendedPrecision) { 1816 // Specify format (uncomment one below to choose) 1817 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1818 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1819 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1820 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1821 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1822 } 1823 if (kEnableExtendedChannels) { 1824 // Specify channel mask (uncomment one below to choose) 1825 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1826 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1827 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1828 } 1829 } 1830 1831 AudioStreamOut *outputStream = NULL; 1832 status_t status = outHwDev->openOutputStream( 1833 &outputStream, 1834 *output, 1835 devices, 1836 flags, 1837 config, 1838 address.string()); 1839 1840 mHardwareStatus = AUDIO_HW_IDLE; 1841 1842 if (status == NO_ERROR) { 1843 1844 PlaybackThread *thread; 1845 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1846 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1847 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1848 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1849 || !isValidPcmSinkFormat(config->format) 1850 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1851 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1852 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1853 } else { 1854 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1855 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1856 } 1857 mPlaybackThreads.add(*output, thread); 1858 return thread; 1859 } 1860 1861 return 0; 1862} 1863 1864status_t AudioFlinger::openOutput(audio_module_handle_t module, 1865 audio_io_handle_t *output, 1866 audio_config_t *config, 1867 audio_devices_t *devices, 1868 const String8& address, 1869 uint32_t *latencyMs, 1870 audio_output_flags_t flags) 1871{ 1872 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1873 module, 1874 (devices != NULL) ? *devices : 0, 1875 config->sample_rate, 1876 config->format, 1877 config->channel_mask, 1878 flags); 1879 1880 if (*devices == AUDIO_DEVICE_NONE) { 1881 return BAD_VALUE; 1882 } 1883 1884 Mutex::Autolock _l(mLock); 1885 1886 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1887 if (thread != 0) { 1888 *latencyMs = thread->latency(); 1889 1890 // notify client processes of the new output creation 1891 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1892 1893 // the first primary output opened designates the primary hw device 1894 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1895 ALOGI("Using module %d has the primary audio interface", module); 1896 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1897 1898 AutoMutex lock(mHardwareLock); 1899 mHardwareStatus = AUDIO_HW_SET_MODE; 1900 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1901 mHardwareStatus = AUDIO_HW_IDLE; 1902 } 1903 return NO_ERROR; 1904 } 1905 1906 return NO_INIT; 1907} 1908 1909audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1910 audio_io_handle_t output2) 1911{ 1912 Mutex::Autolock _l(mLock); 1913 MixerThread *thread1 = checkMixerThread_l(output1); 1914 MixerThread *thread2 = checkMixerThread_l(output2); 1915 1916 if (thread1 == NULL || thread2 == NULL) { 1917 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1918 output2); 1919 return AUDIO_IO_HANDLE_NONE; 1920 } 1921 1922 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1923 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1924 thread->addOutputTrack(thread2); 1925 mPlaybackThreads.add(id, thread); 1926 // notify client processes of the new output creation 1927 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1928 return id; 1929} 1930 1931status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1932{ 1933 return closeOutput_nonvirtual(output); 1934} 1935 1936status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1937{ 1938 // keep strong reference on the playback thread so that 1939 // it is not destroyed while exit() is executed 1940 sp<PlaybackThread> thread; 1941 { 1942 Mutex::Autolock _l(mLock); 1943 thread = checkPlaybackThread_l(output); 1944 if (thread == NULL) { 1945 return BAD_VALUE; 1946 } 1947 1948 ALOGV("closeOutput() %d", output); 1949 1950 if (thread->type() == ThreadBase::MIXER) { 1951 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1952 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1953 DuplicatingThread *dupThread = 1954 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1955 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1956 } 1957 } 1958 } 1959 1960 1961 mPlaybackThreads.removeItem(output); 1962 // save all effects to the default thread 1963 if (mPlaybackThreads.size()) { 1964 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1965 if (dstThread != NULL) { 1966 // audioflinger lock is held here so the acquisition order of thread locks does not 1967 // matter 1968 Mutex::Autolock _dl(dstThread->mLock); 1969 Mutex::Autolock _sl(thread->mLock); 1970 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1971 for (size_t i = 0; i < effectChains.size(); i ++) { 1972 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1973 } 1974 } 1975 } 1976 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1977 ioDesc->mIoHandle = output; 1978 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1979 } 1980 thread->exit(); 1981 // The thread entity (active unit of execution) is no longer running here, 1982 // but the ThreadBase container still exists. 1983 1984 if (!thread->isDuplicating()) { 1985 closeOutputFinish(thread); 1986 } 1987 1988 return NO_ERROR; 1989} 1990 1991void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1992{ 1993 AudioStreamOut *out = thread->clearOutput(); 1994 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1995 // from now on thread->mOutput is NULL 1996 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1997 delete out; 1998} 1999 2000void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 2001{ 2002 mPlaybackThreads.removeItem(thread->mId); 2003 thread->exit(); 2004 closeOutputFinish(thread); 2005} 2006 2007status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2008{ 2009 Mutex::Autolock _l(mLock); 2010 PlaybackThread *thread = checkPlaybackThread_l(output); 2011 2012 if (thread == NULL) { 2013 return BAD_VALUE; 2014 } 2015 2016 ALOGV("suspendOutput() %d", output); 2017 thread->suspend(); 2018 2019 return NO_ERROR; 2020} 2021 2022status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2023{ 2024 Mutex::Autolock _l(mLock); 2025 PlaybackThread *thread = checkPlaybackThread_l(output); 2026 2027 if (thread == NULL) { 2028 return BAD_VALUE; 2029 } 2030 2031 ALOGV("restoreOutput() %d", output); 2032 2033 thread->restore(); 2034 2035 return NO_ERROR; 2036} 2037 2038status_t AudioFlinger::openInput(audio_module_handle_t module, 2039 audio_io_handle_t *input, 2040 audio_config_t *config, 2041 audio_devices_t *devices, 2042 const String8& address, 2043 audio_source_t source, 2044 audio_input_flags_t flags) 2045{ 2046 Mutex::Autolock _l(mLock); 2047 2048 if (*devices == AUDIO_DEVICE_NONE) { 2049 return BAD_VALUE; 2050 } 2051 2052 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2053 2054 if (thread != 0) { 2055 // notify client processes of the new input creation 2056 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2057 return NO_ERROR; 2058 } 2059 return NO_INIT; 2060} 2061 2062sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2063 audio_io_handle_t *input, 2064 audio_config_t *config, 2065 audio_devices_t devices, 2066 const String8& address, 2067 audio_source_t source, 2068 audio_input_flags_t flags) 2069{ 2070 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2071 if (inHwDev == NULL) { 2072 *input = AUDIO_IO_HANDLE_NONE; 2073 return 0; 2074 } 2075 2076 // Audio Policy can request a specific handle for hardware hotword. 2077 // The goal here is not to re-open an already opened input. 2078 // It is to use a pre-assigned I/O handle. 2079 if (*input == AUDIO_IO_HANDLE_NONE) { 2080 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2081 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2082 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2083 return 0; 2084 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2085 // This should not happen in a transient state with current design. 2086 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2087 return 0; 2088 } 2089 2090 audio_config_t halconfig = *config; 2091 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2092 audio_stream_in_t *inStream = NULL; 2093 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2094 &inStream, flags, address.string(), source); 2095 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2096 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2097 inStream, 2098 halconfig.sample_rate, 2099 halconfig.format, 2100 halconfig.channel_mask, 2101 flags, 2102 status, address.string()); 2103 2104 // If the input could not be opened with the requested parameters and we can handle the 2105 // conversion internally, try to open again with the proposed parameters. 2106 if (status == BAD_VALUE && 2107 audio_is_linear_pcm(config->format) && 2108 audio_is_linear_pcm(halconfig.format) && 2109 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2110 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2111 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2112 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2113 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2114 inStream = NULL; 2115 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2116 &inStream, flags, address.string(), source); 2117 // FIXME log this new status; HAL should not propose any further changes 2118 } 2119 2120 if (status == NO_ERROR && inStream != NULL) { 2121 2122#ifdef TEE_SINK 2123 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2124 // or (re-)create if current Pipe is idle and does not match the new format 2125 sp<NBAIO_Sink> teeSink; 2126 enum { 2127 TEE_SINK_NO, // don't copy input 2128 TEE_SINK_NEW, // copy input using a new pipe 2129 TEE_SINK_OLD, // copy input using an existing pipe 2130 } kind; 2131 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2132 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2133 if (!mTeeSinkInputEnabled) { 2134 kind = TEE_SINK_NO; 2135 } else if (!Format_isValid(format)) { 2136 kind = TEE_SINK_NO; 2137 } else if (mRecordTeeSink == 0) { 2138 kind = TEE_SINK_NEW; 2139 } else if (mRecordTeeSink->getStrongCount() != 1) { 2140 kind = TEE_SINK_NO; 2141 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2142 kind = TEE_SINK_OLD; 2143 } else { 2144 kind = TEE_SINK_NEW; 2145 } 2146 switch (kind) { 2147 case TEE_SINK_NEW: { 2148 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2149 size_t numCounterOffers = 0; 2150 const NBAIO_Format offers[1] = {format}; 2151 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2152 ALOG_ASSERT(index == 0); 2153 PipeReader *pipeReader = new PipeReader(*pipe); 2154 numCounterOffers = 0; 2155 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2156 ALOG_ASSERT(index == 0); 2157 mRecordTeeSink = pipe; 2158 mRecordTeeSource = pipeReader; 2159 teeSink = pipe; 2160 } 2161 break; 2162 case TEE_SINK_OLD: 2163 teeSink = mRecordTeeSink; 2164 break; 2165 case TEE_SINK_NO: 2166 default: 2167 break; 2168 } 2169#endif 2170 2171 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2172 2173 // Start record thread 2174 // RecordThread requires both input and output device indication to forward to audio 2175 // pre processing modules 2176 sp<RecordThread> thread = new RecordThread(this, 2177 inputStream, 2178 *input, 2179 primaryOutputDevice_l(), 2180 devices, 2181 mSystemReady 2182#ifdef TEE_SINK 2183 , teeSink 2184#endif 2185 ); 2186 mRecordThreads.add(*input, thread); 2187 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2188 return thread; 2189 } 2190 2191 *input = AUDIO_IO_HANDLE_NONE; 2192 return 0; 2193} 2194 2195status_t AudioFlinger::closeInput(audio_io_handle_t input) 2196{ 2197 return closeInput_nonvirtual(input); 2198} 2199 2200status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2201{ 2202 // keep strong reference on the record thread so that 2203 // it is not destroyed while exit() is executed 2204 sp<RecordThread> thread; 2205 { 2206 Mutex::Autolock _l(mLock); 2207 thread = checkRecordThread_l(input); 2208 if (thread == 0) { 2209 return BAD_VALUE; 2210 } 2211 2212 ALOGV("closeInput() %d", input); 2213 2214 // If we still have effect chains, it means that a client still holds a handle 2215 // on at least one effect. We must either move the chain to an existing thread with the 2216 // same session ID or put it aside in case a new record thread is opened for a 2217 // new capture on the same session 2218 sp<EffectChain> chain; 2219 { 2220 Mutex::Autolock _sl(thread->mLock); 2221 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2222 // Note: maximum one chain per record thread 2223 if (effectChains.size() != 0) { 2224 chain = effectChains[0]; 2225 } 2226 } 2227 if (chain != 0) { 2228 // first check if a record thread is already opened with a client on the same session. 2229 // This should only happen in case of overlap between one thread tear down and the 2230 // creation of its replacement 2231 size_t i; 2232 for (i = 0; i < mRecordThreads.size(); i++) { 2233 sp<RecordThread> t = mRecordThreads.valueAt(i); 2234 if (t == thread) { 2235 continue; 2236 } 2237 if (t->hasAudioSession(chain->sessionId()) != 0) { 2238 Mutex::Autolock _l(t->mLock); 2239 ALOGV("closeInput() found thread %d for effect session %d", 2240 t->id(), chain->sessionId()); 2241 t->addEffectChain_l(chain); 2242 break; 2243 } 2244 } 2245 // put the chain aside if we could not find a record thread with the same session id. 2246 if (i == mRecordThreads.size()) { 2247 putOrphanEffectChain_l(chain); 2248 } 2249 } 2250 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2251 ioDesc->mIoHandle = input; 2252 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2253 mRecordThreads.removeItem(input); 2254 } 2255 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2256 // we have a different lock for notification client 2257 closeInputFinish(thread); 2258 return NO_ERROR; 2259} 2260 2261void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2262{ 2263 thread->exit(); 2264 AudioStreamIn *in = thread->clearInput(); 2265 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2266 // from now on thread->mInput is NULL 2267 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2268 delete in; 2269} 2270 2271void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2272{ 2273 mRecordThreads.removeItem(thread->mId); 2274 closeInputFinish(thread); 2275} 2276 2277status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2278{ 2279 Mutex::Autolock _l(mLock); 2280 ALOGV("invalidateStream() stream %d", stream); 2281 2282 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2283 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2284 thread->invalidateTracks(stream); 2285 } 2286 2287 return NO_ERROR; 2288} 2289 2290 2291audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2292{ 2293 // This is a binder API, so a malicious client could pass in a bad parameter. 2294 // Check for that before calling the internal API nextUniqueId(). 2295 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2296 ALOGE("newAudioUniqueId invalid use %d", use); 2297 return AUDIO_UNIQUE_ID_ALLOCATE; 2298 } 2299 return nextUniqueId(use); 2300} 2301 2302void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2303{ 2304 Mutex::Autolock _l(mLock); 2305 pid_t caller = IPCThreadState::self()->getCallingPid(); 2306 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2307 if (pid != -1 && (caller == getpid_cached)) { 2308 caller = pid; 2309 } 2310 2311 { 2312 Mutex::Autolock _cl(mClientLock); 2313 // Ignore requests received from processes not known as notification client. The request 2314 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2315 // called from a different pid leaving a stale session reference. Also we don't know how 2316 // to clear this reference if the client process dies. 2317 if (mNotificationClients.indexOfKey(caller) < 0) { 2318 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2319 return; 2320 } 2321 } 2322 2323 size_t num = mAudioSessionRefs.size(); 2324 for (size_t i = 0; i< num; i++) { 2325 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2326 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2327 ref->mCnt++; 2328 ALOGV(" incremented refcount to %d", ref->mCnt); 2329 return; 2330 } 2331 } 2332 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2333 ALOGV(" added new entry for %d", audioSession); 2334} 2335 2336void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2337{ 2338 Mutex::Autolock _l(mLock); 2339 pid_t caller = IPCThreadState::self()->getCallingPid(); 2340 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2341 if (pid != -1 && (caller == getpid_cached)) { 2342 caller = pid; 2343 } 2344 size_t num = mAudioSessionRefs.size(); 2345 for (size_t i = 0; i< num; i++) { 2346 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2347 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2348 ref->mCnt--; 2349 ALOGV(" decremented refcount to %d", ref->mCnt); 2350 if (ref->mCnt == 0) { 2351 mAudioSessionRefs.removeAt(i); 2352 delete ref; 2353 purgeStaleEffects_l(); 2354 } 2355 return; 2356 } 2357 } 2358 // If the caller is mediaserver it is likely that the session being released was acquired 2359 // on behalf of a process not in notification clients and we ignore the warning. 2360 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2361} 2362 2363void AudioFlinger::purgeStaleEffects_l() { 2364 2365 ALOGV("purging stale effects"); 2366 2367 Vector< sp<EffectChain> > chains; 2368 2369 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2370 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2371 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2372 sp<EffectChain> ec = t->mEffectChains[j]; 2373 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2374 chains.push(ec); 2375 } 2376 } 2377 } 2378 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2379 sp<RecordThread> t = mRecordThreads.valueAt(i); 2380 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2381 sp<EffectChain> ec = t->mEffectChains[j]; 2382 chains.push(ec); 2383 } 2384 } 2385 2386 for (size_t i = 0; i < chains.size(); i++) { 2387 sp<EffectChain> ec = chains[i]; 2388 int sessionid = ec->sessionId(); 2389 sp<ThreadBase> t = ec->mThread.promote(); 2390 if (t == 0) { 2391 continue; 2392 } 2393 size_t numsessionrefs = mAudioSessionRefs.size(); 2394 bool found = false; 2395 for (size_t k = 0; k < numsessionrefs; k++) { 2396 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2397 if (ref->mSessionid == sessionid) { 2398 ALOGV(" session %d still exists for %d with %d refs", 2399 sessionid, ref->mPid, ref->mCnt); 2400 found = true; 2401 break; 2402 } 2403 } 2404 if (!found) { 2405 Mutex::Autolock _l(t->mLock); 2406 // remove all effects from the chain 2407 while (ec->mEffects.size()) { 2408 sp<EffectModule> effect = ec->mEffects[0]; 2409 effect->unPin(); 2410 t->removeEffect_l(effect); 2411 if (effect->purgeHandles()) { 2412 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2413 } 2414 AudioSystem::unregisterEffect(effect->id()); 2415 } 2416 } 2417 } 2418 return; 2419} 2420 2421// checkThread_l() must be called with AudioFlinger::mLock held 2422AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2423{ 2424 ThreadBase *thread = NULL; 2425 switch (audio_unique_id_get_use(ioHandle)) { 2426 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2427 thread = checkPlaybackThread_l(ioHandle); 2428 break; 2429 case AUDIO_UNIQUE_ID_USE_INPUT: 2430 thread = checkRecordThread_l(ioHandle); 2431 break; 2432 default: 2433 break; 2434 } 2435 return thread; 2436} 2437 2438// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2439AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2440{ 2441 return mPlaybackThreads.valueFor(output).get(); 2442} 2443 2444// checkMixerThread_l() must be called with AudioFlinger::mLock held 2445AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2446{ 2447 PlaybackThread *thread = checkPlaybackThread_l(output); 2448 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2449} 2450 2451// checkRecordThread_l() must be called with AudioFlinger::mLock held 2452AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2453{ 2454 return mRecordThreads.valueFor(input).get(); 2455} 2456 2457audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2458{ 2459 // This is the internal API, so it is OK to assert on bad parameter. 2460 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2461 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2462 for (int retry = 0; retry < maxRetries; retry++) { 2463 // The cast allows wraparound from max positive to min negative instead of abort 2464 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2465 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2466 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2467 // allow wrap by skipping 0 and -1 for session ids 2468 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2469 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2470 return (audio_unique_id_t) (base | use); 2471 } 2472 } 2473 // We have no way of recovering from wraparound 2474 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2475 // TODO Use a floor after wraparound. This may need a mutex. 2476} 2477 2478AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2479{ 2480 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2481 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2482 if(thread->isDuplicating()) { 2483 continue; 2484 } 2485 AudioStreamOut *output = thread->getOutput(); 2486 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2487 return thread; 2488 } 2489 } 2490 return NULL; 2491} 2492 2493audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2494{ 2495 PlaybackThread *thread = primaryPlaybackThread_l(); 2496 2497 if (thread == NULL) { 2498 return 0; 2499 } 2500 2501 return thread->outDevice(); 2502} 2503 2504sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2505 audio_session_t triggerSession, 2506 audio_session_t listenerSession, 2507 sync_event_callback_t callBack, 2508 wp<RefBase> cookie) 2509{ 2510 Mutex::Autolock _l(mLock); 2511 2512 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2513 status_t playStatus = NAME_NOT_FOUND; 2514 status_t recStatus = NAME_NOT_FOUND; 2515 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2516 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2517 if (playStatus == NO_ERROR) { 2518 return event; 2519 } 2520 } 2521 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2522 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2523 if (recStatus == NO_ERROR) { 2524 return event; 2525 } 2526 } 2527 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2528 mPendingSyncEvents.add(event); 2529 } else { 2530 ALOGV("createSyncEvent() invalid event %d", event->type()); 2531 event.clear(); 2532 } 2533 return event; 2534} 2535 2536// ---------------------------------------------------------------------------- 2537// Effect management 2538// ---------------------------------------------------------------------------- 2539 2540 2541status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2542{ 2543 Mutex::Autolock _l(mLock); 2544 return EffectQueryNumberEffects(numEffects); 2545} 2546 2547status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2548{ 2549 Mutex::Autolock _l(mLock); 2550 return EffectQueryEffect(index, descriptor); 2551} 2552 2553status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2554 effect_descriptor_t *descriptor) const 2555{ 2556 Mutex::Autolock _l(mLock); 2557 return EffectGetDescriptor(pUuid, descriptor); 2558} 2559 2560 2561sp<IEffect> AudioFlinger::createEffect( 2562 effect_descriptor_t *pDesc, 2563 const sp<IEffectClient>& effectClient, 2564 int32_t priority, 2565 audio_io_handle_t io, 2566 audio_session_t sessionId, 2567 const String16& opPackageName, 2568 status_t *status, 2569 int *id, 2570 int *enabled) 2571{ 2572 status_t lStatus = NO_ERROR; 2573 sp<EffectHandle> handle; 2574 effect_descriptor_t desc; 2575 2576 pid_t pid = IPCThreadState::self()->getCallingPid(); 2577 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2578 pid, effectClient.get(), priority, sessionId, io); 2579 2580 if (pDesc == NULL) { 2581 lStatus = BAD_VALUE; 2582 goto Exit; 2583 } 2584 2585 // check audio settings permission for global effects 2586 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2587 lStatus = PERMISSION_DENIED; 2588 goto Exit; 2589 } 2590 2591 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2592 // that can only be created by audio policy manager (running in same process) 2593 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2594 lStatus = PERMISSION_DENIED; 2595 goto Exit; 2596 } 2597 2598 { 2599 if (!EffectIsNullUuid(&pDesc->uuid)) { 2600 // if uuid is specified, request effect descriptor 2601 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2602 if (lStatus < 0) { 2603 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2604 goto Exit; 2605 } 2606 } else { 2607 // if uuid is not specified, look for an available implementation 2608 // of the required type in effect factory 2609 if (EffectIsNullUuid(&pDesc->type)) { 2610 ALOGW("createEffect() no effect type"); 2611 lStatus = BAD_VALUE; 2612 goto Exit; 2613 } 2614 uint32_t numEffects = 0; 2615 effect_descriptor_t d; 2616 d.flags = 0; // prevent compiler warning 2617 bool found = false; 2618 2619 lStatus = EffectQueryNumberEffects(&numEffects); 2620 if (lStatus < 0) { 2621 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2622 goto Exit; 2623 } 2624 for (uint32_t i = 0; i < numEffects; i++) { 2625 lStatus = EffectQueryEffect(i, &desc); 2626 if (lStatus < 0) { 2627 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2628 continue; 2629 } 2630 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2631 // If matching type found save effect descriptor. If the session is 2632 // 0 and the effect is not auxiliary, continue enumeration in case 2633 // an auxiliary version of this effect type is available 2634 found = true; 2635 d = desc; 2636 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2637 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2638 break; 2639 } 2640 } 2641 } 2642 if (!found) { 2643 lStatus = BAD_VALUE; 2644 ALOGW("createEffect() effect not found"); 2645 goto Exit; 2646 } 2647 // For same effect type, chose auxiliary version over insert version if 2648 // connect to output mix (Compliance to OpenSL ES) 2649 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2650 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2651 desc = d; 2652 } 2653 } 2654 2655 // Do not allow auxiliary effects on a session different from 0 (output mix) 2656 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2657 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2658 lStatus = INVALID_OPERATION; 2659 goto Exit; 2660 } 2661 2662 // check recording permission for visualizer 2663 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2664 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2665 lStatus = PERMISSION_DENIED; 2666 goto Exit; 2667 } 2668 2669 // return effect descriptor 2670 *pDesc = desc; 2671 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2672 // if the output returned by getOutputForEffect() is removed before we lock the 2673 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2674 // and we will exit safely 2675 io = AudioSystem::getOutputForEffect(&desc); 2676 ALOGV("createEffect got output %d", io); 2677 } 2678 2679 Mutex::Autolock _l(mLock); 2680 2681 // If output is not specified try to find a matching audio session ID in one of the 2682 // output threads. 2683 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2684 // because of code checking output when entering the function. 2685 // Note: io is never 0 when creating an effect on an input 2686 if (io == AUDIO_IO_HANDLE_NONE) { 2687 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2688 // output must be specified by AudioPolicyManager when using session 2689 // AUDIO_SESSION_OUTPUT_STAGE 2690 lStatus = BAD_VALUE; 2691 goto Exit; 2692 } 2693 // look for the thread where the specified audio session is present 2694 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2695 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2696 io = mPlaybackThreads.keyAt(i); 2697 break; 2698 } 2699 } 2700 if (io == 0) { 2701 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2702 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2703 io = mRecordThreads.keyAt(i); 2704 break; 2705 } 2706 } 2707 } 2708 // If no output thread contains the requested session ID, default to 2709 // first output. The effect chain will be moved to the correct output 2710 // thread when a track with the same session ID is created 2711 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2712 io = mPlaybackThreads.keyAt(0); 2713 } 2714 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2715 } 2716 ThreadBase *thread = checkRecordThread_l(io); 2717 if (thread == NULL) { 2718 thread = checkPlaybackThread_l(io); 2719 if (thread == NULL) { 2720 ALOGE("createEffect() unknown output thread"); 2721 lStatus = BAD_VALUE; 2722 goto Exit; 2723 } 2724 } else { 2725 // Check if one effect chain was awaiting for an effect to be created on this 2726 // session and used it instead of creating a new one. 2727 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2728 if (chain != 0) { 2729 Mutex::Autolock _l(thread->mLock); 2730 thread->addEffectChain_l(chain); 2731 } 2732 } 2733 2734 sp<Client> client = registerPid(pid); 2735 2736 // create effect on selected output thread 2737 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2738 &desc, enabled, &lStatus); 2739 if (handle != 0 && id != NULL) { 2740 *id = handle->id(); 2741 } 2742 if (handle == 0) { 2743 // remove local strong reference to Client with mClientLock held 2744 Mutex::Autolock _cl(mClientLock); 2745 client.clear(); 2746 } 2747 } 2748 2749Exit: 2750 *status = lStatus; 2751 return handle; 2752} 2753 2754status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2755 audio_io_handle_t dstOutput) 2756{ 2757 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2758 sessionId, srcOutput, dstOutput); 2759 Mutex::Autolock _l(mLock); 2760 if (srcOutput == dstOutput) { 2761 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2762 return NO_ERROR; 2763 } 2764 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2765 if (srcThread == NULL) { 2766 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2767 return BAD_VALUE; 2768 } 2769 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2770 if (dstThread == NULL) { 2771 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2772 return BAD_VALUE; 2773 } 2774 2775 Mutex::Autolock _dl(dstThread->mLock); 2776 Mutex::Autolock _sl(srcThread->mLock); 2777 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2778} 2779 2780// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2781status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2782 AudioFlinger::PlaybackThread *srcThread, 2783 AudioFlinger::PlaybackThread *dstThread, 2784 bool reRegister) 2785{ 2786 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2787 sessionId, srcThread, dstThread); 2788 2789 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2790 if (chain == 0) { 2791 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2792 sessionId, srcThread); 2793 return INVALID_OPERATION; 2794 } 2795 2796 // Check whether the destination thread has a channel count of FCC_2, which is 2797 // currently required for (most) effects. Prevent moving the effect chain here rather 2798 // than disabling the addEffect_l() call in dstThread below. 2799 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2800 dstThread->mChannelCount != FCC_2) { 2801 ALOGW("moveEffectChain_l() effect chain failed because" 2802 " destination thread %p channel count(%u) != %u", 2803 dstThread, dstThread->mChannelCount, FCC_2); 2804 return INVALID_OPERATION; 2805 } 2806 2807 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2808 // so that a new chain is created with correct parameters when first effect is added. This is 2809 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2810 // removed. 2811 srcThread->removeEffectChain_l(chain); 2812 2813 // transfer all effects one by one so that new effect chain is created on new thread with 2814 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2815 sp<EffectChain> dstChain; 2816 uint32_t strategy = 0; // prevent compiler warning 2817 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2818 Vector< sp<EffectModule> > removed; 2819 status_t status = NO_ERROR; 2820 while (effect != 0) { 2821 srcThread->removeEffect_l(effect); 2822 removed.add(effect); 2823 status = dstThread->addEffect_l(effect); 2824 if (status != NO_ERROR) { 2825 break; 2826 } 2827 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2828 if (effect->state() == EffectModule::ACTIVE || 2829 effect->state() == EffectModule::STOPPING) { 2830 effect->start(); 2831 } 2832 // if the move request is not received from audio policy manager, the effect must be 2833 // re-registered with the new strategy and output 2834 if (dstChain == 0) { 2835 dstChain = effect->chain().promote(); 2836 if (dstChain == 0) { 2837 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2838 status = NO_INIT; 2839 break; 2840 } 2841 strategy = dstChain->strategy(); 2842 } 2843 if (reRegister) { 2844 AudioSystem::unregisterEffect(effect->id()); 2845 AudioSystem::registerEffect(&effect->desc(), 2846 dstThread->id(), 2847 strategy, 2848 sessionId, 2849 effect->id()); 2850 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2851 } 2852 effect = chain->getEffectFromId_l(0); 2853 } 2854 2855 if (status != NO_ERROR) { 2856 for (size_t i = 0; i < removed.size(); i++) { 2857 srcThread->addEffect_l(removed[i]); 2858 if (dstChain != 0 && reRegister) { 2859 AudioSystem::unregisterEffect(removed[i]->id()); 2860 AudioSystem::registerEffect(&removed[i]->desc(), 2861 srcThread->id(), 2862 strategy, 2863 sessionId, 2864 removed[i]->id()); 2865 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2866 } 2867 } 2868 } 2869 2870 return status; 2871} 2872 2873bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2874{ 2875 if (mGlobalEffectEnableTime != 0 && 2876 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2877 return true; 2878 } 2879 2880 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2881 sp<EffectChain> ec = 2882 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2883 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2884 return true; 2885 } 2886 } 2887 return false; 2888} 2889 2890void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2891{ 2892 Mutex::Autolock _l(mLock); 2893 2894 mGlobalEffectEnableTime = systemTime(); 2895 2896 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2897 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2898 if (t->mType == ThreadBase::OFFLOAD) { 2899 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2900 } 2901 } 2902 2903} 2904 2905status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2906{ 2907 audio_session_t session = chain->sessionId(); 2908 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2909 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2910 if (index >= 0) { 2911 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2912 return ALREADY_EXISTS; 2913 } 2914 mOrphanEffectChains.add(session, chain); 2915 return NO_ERROR; 2916} 2917 2918sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2919{ 2920 sp<EffectChain> chain; 2921 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2922 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2923 if (index >= 0) { 2924 chain = mOrphanEffectChains.valueAt(index); 2925 mOrphanEffectChains.removeItemsAt(index); 2926 } 2927 return chain; 2928} 2929 2930bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2931{ 2932 Mutex::Autolock _l(mLock); 2933 audio_session_t session = effect->sessionId(); 2934 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2935 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2936 if (index >= 0) { 2937 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2938 if (chain->removeEffect_l(effect) == 0) { 2939 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2940 mOrphanEffectChains.removeItemsAt(index); 2941 } 2942 return true; 2943 } 2944 return false; 2945} 2946 2947 2948struct Entry { 2949#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2950 char mFileName[TEE_MAX_FILENAME]; 2951}; 2952 2953int comparEntry(const void *p1, const void *p2) 2954{ 2955 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2956} 2957 2958#ifdef TEE_SINK 2959void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2960{ 2961 NBAIO_Source *teeSource = source.get(); 2962 if (teeSource != NULL) { 2963 // .wav rotation 2964 // There is a benign race condition if 2 threads call this simultaneously. 2965 // They would both traverse the directory, but the result would simply be 2966 // failures at unlink() which are ignored. It's also unlikely since 2967 // normally dumpsys is only done by bugreport or from the command line. 2968 char teePath[32+256]; 2969 strcpy(teePath, "/data/misc/audioserver"); 2970 size_t teePathLen = strlen(teePath); 2971 DIR *dir = opendir(teePath); 2972 teePath[teePathLen++] = '/'; 2973 if (dir != NULL) { 2974#define TEE_MAX_SORT 20 // number of entries to sort 2975#define TEE_MAX_KEEP 10 // number of entries to keep 2976 struct Entry entries[TEE_MAX_SORT]; 2977 size_t entryCount = 0; 2978 while (entryCount < TEE_MAX_SORT) { 2979 struct dirent de; 2980 struct dirent *result = NULL; 2981 int rc = readdir_r(dir, &de, &result); 2982 if (rc != 0) { 2983 ALOGW("readdir_r failed %d", rc); 2984 break; 2985 } 2986 if (result == NULL) { 2987 break; 2988 } 2989 if (result != &de) { 2990 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2991 break; 2992 } 2993 // ignore non .wav file entries 2994 size_t nameLen = strlen(de.d_name); 2995 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2996 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2997 continue; 2998 } 2999 strcpy(entries[entryCount++].mFileName, de.d_name); 3000 } 3001 (void) closedir(dir); 3002 if (entryCount > TEE_MAX_KEEP) { 3003 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3004 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3005 strcpy(&teePath[teePathLen], entries[i].mFileName); 3006 (void) unlink(teePath); 3007 } 3008 } 3009 } else { 3010 if (fd >= 0) { 3011 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 3012 strerror(errno)); 3013 } 3014 } 3015 char teeTime[16]; 3016 struct timeval tv; 3017 gettimeofday(&tv, NULL); 3018 struct tm tm; 3019 localtime_r(&tv.tv_sec, &tm); 3020 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3021 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3022 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3023 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3024 if (teeFd >= 0) { 3025 // FIXME use libsndfile 3026 char wavHeader[44]; 3027 memcpy(wavHeader, 3028 "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", 3029 sizeof(wavHeader)); 3030 NBAIO_Format format = teeSource->format(); 3031 unsigned channelCount = Format_channelCount(format); 3032 uint32_t sampleRate = Format_sampleRate(format); 3033 size_t frameSize = Format_frameSize(format); 3034 wavHeader[22] = channelCount; // number of channels 3035 wavHeader[24] = sampleRate; // sample rate 3036 wavHeader[25] = sampleRate >> 8; 3037 wavHeader[32] = frameSize; // block alignment 3038 wavHeader[33] = frameSize >> 8; 3039 write(teeFd, wavHeader, sizeof(wavHeader)); 3040 size_t total = 0; 3041 bool firstRead = true; 3042#define TEE_SINK_READ 1024 // frames per I/O operation 3043 void *buffer = malloc(TEE_SINK_READ * frameSize); 3044 for (;;) { 3045 size_t count = TEE_SINK_READ; 3046 ssize_t actual = teeSource->read(buffer, count); 3047 bool wasFirstRead = firstRead; 3048 firstRead = false; 3049 if (actual <= 0) { 3050 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3051 continue; 3052 } 3053 break; 3054 } 3055 ALOG_ASSERT(actual <= (ssize_t)count); 3056 write(teeFd, buffer, actual * frameSize); 3057 total += actual; 3058 } 3059 free(buffer); 3060 lseek(teeFd, (off_t) 4, SEEK_SET); 3061 uint32_t temp = 44 + total * frameSize - 8; 3062 // FIXME not big-endian safe 3063 write(teeFd, &temp, sizeof(temp)); 3064 lseek(teeFd, (off_t) 40, SEEK_SET); 3065 temp = total * frameSize; 3066 // FIXME not big-endian safe 3067 write(teeFd, &temp, sizeof(temp)); 3068 close(teeFd); 3069 if (fd >= 0) { 3070 dprintf(fd, "tee copied to %s\n", teePath); 3071 } 3072 } else { 3073 if (fd >= 0) { 3074 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3075 } 3076 } 3077 } 3078} 3079#endif 3080 3081// ---------------------------------------------------------------------------- 3082 3083status_t AudioFlinger::onTransact( 3084 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3085{ 3086 return BnAudioFlinger::onTransact(code, data, reply, flags); 3087} 3088 3089} // namespace android 3090