AudioFlinger.cpp revision 329f6511ee4e03a4605c70bbda8d3a96d2544884
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/EffectsFactoryApi.h> 49#include <audio_effects/effect_visualizer.h> 50#include <audio_effects/effect_ns.h> 51#include <audio_effects/effect_aec.h> 52 53#include <audio_utils/primitives.h> 54 55#include <powermanager/PowerManager.h> 56 57#include <common_time/cc_helper.h> 58 59#include <media/IMediaLogService.h> 60 61#include <media/nbaio/Pipe.h> 62#include <media/nbaio/PipeReader.h> 63#include <media/AudioParameter.h> 64#include <private/android_filesystem_config.h> 65 66// ---------------------------------------------------------------------------- 67 68// Note: the following macro is used for extremely verbose logging message. In 69// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 70// 0; but one side effect of this is to turn all LOGV's as well. Some messages 71// are so verbose that we want to suppress them even when we have ALOG_ASSERT 72// turned on. Do not uncomment the #def below unless you really know what you 73// are doing and want to see all of the extremely verbose messages. 74//#define VERY_VERY_VERBOSE_LOGGING 75#ifdef VERY_VERY_VERBOSE_LOGGING 76#define ALOGVV ALOGV 77#else 78#define ALOGVV(a...) do { } while(0) 79#endif 80 81namespace android { 82 83static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 84static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 85static const char kClientLockedString[] = "Client lock is taken\n"; 86 87 88nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 89 90uint32_t AudioFlinger::mScreenState; 91 92#ifdef TEE_SINK 93bool AudioFlinger::mTeeSinkInputEnabled = false; 94bool AudioFlinger::mTeeSinkOutputEnabled = false; 95bool AudioFlinger::mTeeSinkTrackEnabled = false; 96 97size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 98size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 99size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 100#endif 101 102// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 103// we define a minimum time during which a global effect is considered enabled. 104static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 105 106// ---------------------------------------------------------------------------- 107 108const char *formatToString(audio_format_t format) { 109 switch (format & AUDIO_FORMAT_MAIN_MASK) { 110 case AUDIO_FORMAT_PCM: 111 switch (format) { 112 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 113 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 114 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 115 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 116 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 117 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 118 default: 119 break; 120 } 121 break; 122 case AUDIO_FORMAT_MP3: return "mp3"; 123 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 124 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 125 case AUDIO_FORMAT_AAC: return "aac"; 126 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 127 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 128 case AUDIO_FORMAT_VORBIS: return "vorbis"; 129 case AUDIO_FORMAT_OPUS: return "opus"; 130 case AUDIO_FORMAT_AC3: return "ac-3"; 131 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 132 default: 133 break; 134 } 135 return "unknown"; 136} 137 138static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 139{ 140 const hw_module_t *mod; 141 int rc; 142 143 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 144 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 145 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 146 if (rc) { 147 goto out; 148 } 149 rc = audio_hw_device_open(mod, dev); 150 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 156 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 157 rc = BAD_VALUE; 158 goto out; 159 } 160 return 0; 161 162out: 163 *dev = NULL; 164 return rc; 165} 166 167// ---------------------------------------------------------------------------- 168 169AudioFlinger::AudioFlinger() 170 : BnAudioFlinger(), 171 mPrimaryHardwareDev(NULL), 172 mAudioHwDevs(NULL), 173 mHardwareStatus(AUDIO_HW_IDLE), 174 mMasterVolume(1.0f), 175 mMasterMute(false), 176 mNextUniqueId(1), 177 mMode(AUDIO_MODE_INVALID), 178 mBtNrecIsOff(false), 179 mIsLowRamDevice(true), 180 mIsDeviceTypeKnown(false), 181 mGlobalEffectEnableTime(0), 182 mPrimaryOutputSampleRate(0) 183{ 184 getpid_cached = getpid(); 185 char value[PROPERTY_VALUE_MAX]; 186 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 187 if (doLog) { 188 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", MemoryHeapBase::READ_ONLY); 189 } 190 191#ifdef TEE_SINK 192 (void) property_get("ro.debuggable", value, "0"); 193 int debuggable = atoi(value); 194 int teeEnabled = 0; 195 if (debuggable) { 196 (void) property_get("af.tee", value, "0"); 197 teeEnabled = atoi(value); 198 } 199 // FIXME symbolic constants here 200 if (teeEnabled & 1) { 201 mTeeSinkInputEnabled = true; 202 } 203 if (teeEnabled & 2) { 204 mTeeSinkOutputEnabled = true; 205 } 206 if (teeEnabled & 4) { 207 mTeeSinkTrackEnabled = true; 208 } 209#endif 210} 211 212void AudioFlinger::onFirstRef() 213{ 214 int rc = 0; 215 216 Mutex::Autolock _l(mLock); 217 218 /* TODO: move all this work into an Init() function */ 219 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 220 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 221 uint32_t int_val; 222 if (1 == sscanf(val_str, "%u", &int_val)) { 223 mStandbyTimeInNsecs = milliseconds(int_val); 224 ALOGI("Using %u mSec as standby time.", int_val); 225 } else { 226 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 227 ALOGI("Using default %u mSec as standby time.", 228 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 229 } 230 } 231 232 mPatchPanel = new PatchPanel(this); 233 234 mMode = AUDIO_MODE_NORMAL; 235} 236 237AudioFlinger::~AudioFlinger() 238{ 239 while (!mRecordThreads.isEmpty()) { 240 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 241 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 242 } 243 while (!mPlaybackThreads.isEmpty()) { 244 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 245 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 246 } 247 248 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 249 // no mHardwareLock needed, as there are no other references to this 250 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 251 delete mAudioHwDevs.valueAt(i); 252 } 253 254 // Tell media.log service about any old writers that still need to be unregistered 255 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 256 if (binder != 0) { 257 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 258 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 259 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 260 mUnregisteredWriters.pop(); 261 mediaLogService->unregisterWriter(iMemory); 262 } 263 } 264 265} 266 267static const char * const audio_interfaces[] = { 268 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 269 AUDIO_HARDWARE_MODULE_ID_A2DP, 270 AUDIO_HARDWARE_MODULE_ID_USB, 271}; 272#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 273 274AudioFlinger::AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 275 audio_module_handle_t module, 276 audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 // then try to find a module supporting the requested device. 286 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 287 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 288 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 289 if ((dev->get_supported_devices != NULL) && 290 (dev->get_supported_devices(dev) & devices) == devices) 291 return audioHwDevice; 292 } 293 } else { 294 // check a match for the requested module handle 295 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 296 if (audioHwDevice != NULL) { 297 return audioHwDevice; 298 } 299 } 300 301 return NULL; 302} 303 304void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 305{ 306 const size_t SIZE = 256; 307 char buffer[SIZE]; 308 String8 result; 309 310 result.append("Clients:\n"); 311 for (size_t i = 0; i < mClients.size(); ++i) { 312 sp<Client> client = mClients.valueAt(i).promote(); 313 if (client != 0) { 314 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 315 result.append(buffer); 316 } 317 } 318 319 result.append("Notification Clients:\n"); 320 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 321 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 322 result.append(buffer); 323 } 324 325 result.append("Global session refs:\n"); 326 result.append(" session pid count\n"); 327 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 328 AudioSessionRef *r = mAudioSessionRefs[i]; 329 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 330 result.append(buffer); 331 } 332 write(fd, result.string(), result.size()); 333} 334 335 336void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 337{ 338 const size_t SIZE = 256; 339 char buffer[SIZE]; 340 String8 result; 341 hardware_call_state hardwareStatus = mHardwareStatus; 342 343 snprintf(buffer, SIZE, "Hardware status: %d\n" 344 "Standby Time mSec: %u\n", 345 hardwareStatus, 346 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 347 result.append(buffer); 348 write(fd, result.string(), result.size()); 349} 350 351void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 352{ 353 const size_t SIZE = 256; 354 char buffer[SIZE]; 355 String8 result; 356 snprintf(buffer, SIZE, "Permission Denial: " 357 "can't dump AudioFlinger from pid=%d, uid=%d\n", 358 IPCThreadState::self()->getCallingPid(), 359 IPCThreadState::self()->getCallingUid()); 360 result.append(buffer); 361 write(fd, result.string(), result.size()); 362} 363 364bool AudioFlinger::dumpTryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = dumpTryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = dumpTryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 bool clientLocked = dumpTryLock(mClientLock); 400 if (!clientLocked) { 401 String8 result(kClientLockedString); 402 write(fd, result.string(), result.size()); 403 } 404 dumpClients(fd, args); 405 if (clientLocked) { 406 mClientLock.unlock(); 407 } 408 409 dumpInternals(fd, args); 410 411 // dump playback threads 412 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 413 mPlaybackThreads.valueAt(i)->dump(fd, args); 414 } 415 416 // dump record threads 417 for (size_t i = 0; i < mRecordThreads.size(); i++) { 418 mRecordThreads.valueAt(i)->dump(fd, args); 419 } 420 421 // dump all hardware devs 422 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 423 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 424 dev->dump(dev, fd); 425 } 426 427#ifdef TEE_SINK 428 // dump the serially shared record tee sink 429 if (mRecordTeeSource != 0) { 430 dumpTee(fd, mRecordTeeSource); 431 } 432#endif 433 434 if (locked) { 435 mLock.unlock(); 436 } 437 438 // append a copy of media.log here by forwarding fd to it, but don't attempt 439 // to lookup the service if it's not running, as it will block for a second 440 if (mLogMemoryDealer != 0) { 441 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 442 if (binder != 0) { 443 dprintf(fd, "\nmedia.log:\n"); 444 Vector<String16> args; 445 binder->dump(fd, args); 446 } 447 } 448 } 449 return NO_ERROR; 450} 451 452sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 453{ 454 Mutex::Autolock _cl(mClientLock); 455 // If pid is already in the mClients wp<> map, then use that entry 456 // (for which promote() is always != 0), otherwise create a new entry and Client. 457 sp<Client> client = mClients.valueFor(pid).promote(); 458 if (client == 0) { 459 client = new Client(this, pid); 460 mClients.add(pid, client); 461 } 462 463 return client; 464} 465 466sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 467{ 468 // If there is no memory allocated for logs, return a dummy writer that does nothing 469 if (mLogMemoryDealer == 0) { 470 return new NBLog::Writer(); 471 } 472 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 473 // Similarly if we can't contact the media.log service, also return a dummy writer 474 if (binder == 0) { 475 return new NBLog::Writer(); 476 } 477 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 478 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 479 // If allocation fails, consult the vector of previously unregistered writers 480 // and garbage-collect one or more them until an allocation succeeds 481 if (shared == 0) { 482 Mutex::Autolock _l(mUnregisteredWritersLock); 483 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 484 { 485 // Pick the oldest stale writer to garbage-collect 486 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 487 mUnregisteredWriters.removeAt(0); 488 mediaLogService->unregisterWriter(iMemory); 489 // Now the media.log remote reference to IMemory is gone. When our last local 490 // reference to IMemory also drops to zero at end of this block, 491 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 492 } 493 // Re-attempt the allocation 494 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 495 if (shared != 0) { 496 goto success; 497 } 498 } 499 // Even after garbage-collecting all old writers, there is still not enough memory, 500 // so return a dummy writer 501 return new NBLog::Writer(); 502 } 503success: 504 mediaLogService->registerWriter(shared, size, name); 505 return new NBLog::Writer(size, shared); 506} 507 508void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 509{ 510 if (writer == 0) { 511 return; 512 } 513 sp<IMemory> iMemory(writer->getIMemory()); 514 if (iMemory == 0) { 515 return; 516 } 517 // Rather than removing the writer immediately, append it to a queue of old writers to 518 // be garbage-collected later. This allows us to continue to view old logs for a while. 519 Mutex::Autolock _l(mUnregisteredWritersLock); 520 mUnregisteredWriters.push(writer); 521} 522 523// IAudioFlinger interface 524 525 526sp<IAudioTrack> AudioFlinger::createTrack( 527 audio_stream_type_t streamType, 528 uint32_t sampleRate, 529 audio_format_t format, 530 audio_channel_mask_t channelMask, 531 size_t *frameCount, 532 IAudioFlinger::track_flags_t *flags, 533 const sp<IMemory>& sharedBuffer, 534 audio_io_handle_t output, 535 pid_t tid, 536 int *sessionId, 537 int clientUid, 538 status_t *status) 539{ 540 sp<PlaybackThread::Track> track; 541 sp<TrackHandle> trackHandle; 542 sp<Client> client; 543 status_t lStatus; 544 int lSessionId; 545 546 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 547 // but if someone uses binder directly they could bypass that and cause us to crash 548 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 549 ALOGE("createTrack() invalid stream type %d", streamType); 550 lStatus = BAD_VALUE; 551 goto Exit; 552 } 553 554 // further sample rate checks are performed by createTrack_l() depending on the thread type 555 if (sampleRate == 0) { 556 ALOGE("createTrack() invalid sample rate %u", sampleRate); 557 lStatus = BAD_VALUE; 558 goto Exit; 559 } 560 561 // further channel mask checks are performed by createTrack_l() depending on the thread type 562 if (!audio_is_output_channel(channelMask)) { 563 ALOGE("createTrack() invalid channel mask %#x", channelMask); 564 lStatus = BAD_VALUE; 565 goto Exit; 566 } 567 568 // further format checks are performed by createTrack_l() depending on the thread type 569 if (!audio_is_valid_format(format)) { 570 ALOGE("createTrack() invalid format %#x", format); 571 lStatus = BAD_VALUE; 572 goto Exit; 573 } 574 575 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 576 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 { 582 Mutex::Autolock _l(mLock); 583 PlaybackThread *thread = checkPlaybackThread_l(output); 584 if (thread == NULL) { 585 ALOGE("no playback thread found for output handle %d", output); 586 lStatus = BAD_VALUE; 587 goto Exit; 588 } 589 590 pid_t pid = IPCThreadState::self()->getCallingPid(); 591 client = registerPid(pid); 592 593 PlaybackThread *effectThread = NULL; 594 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 595 lSessionId = *sessionId; 596 // check if an effect chain with the same session ID is present on another 597 // output thread and move it here. 598 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 599 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 600 if (mPlaybackThreads.keyAt(i) != output) { 601 uint32_t sessions = t->hasAudioSession(lSessionId); 602 if (sessions & PlaybackThread::EFFECT_SESSION) { 603 effectThread = t.get(); 604 break; 605 } 606 } 607 } 608 } else { 609 // if no audio session id is provided, create one here 610 lSessionId = nextUniqueId(); 611 if (sessionId != NULL) { 612 *sessionId = lSessionId; 613 } 614 } 615 ALOGV("createTrack() lSessionId: %d", lSessionId); 616 617 track = thread->createTrack_l(client, streamType, sampleRate, format, 618 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 619 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 620 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 621 622 // move effect chain to this output thread if an effect on same session was waiting 623 // for a track to be created 624 if (lStatus == NO_ERROR && effectThread != NULL) { 625 // no risk of deadlock because AudioFlinger::mLock is held 626 Mutex::Autolock _dl(thread->mLock); 627 Mutex::Autolock _sl(effectThread->mLock); 628 moveEffectChain_l(lSessionId, effectThread, thread, true); 629 } 630 631 // Look for sync events awaiting for a session to be used. 632 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 633 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 634 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 635 if (lStatus == NO_ERROR) { 636 (void) track->setSyncEvent(mPendingSyncEvents[i]); 637 } else { 638 mPendingSyncEvents[i]->cancel(); 639 } 640 mPendingSyncEvents.removeAt(i); 641 i--; 642 } 643 } 644 } 645 646 } 647 648 if (lStatus != NO_ERROR) { 649 // remove local strong reference to Client before deleting the Track so that the 650 // Client destructor is called by the TrackBase destructor with mClientLock held 651 // Don't hold mClientLock when releasing the reference on the track as the 652 // destructor will acquire it. 653 { 654 Mutex::Autolock _cl(mClientLock); 655 client.clear(); 656 } 657 track.clear(); 658 goto Exit; 659 } 660 661 // return handle to client 662 trackHandle = new TrackHandle(track); 663 664Exit: 665 *status = lStatus; 666 return trackHandle; 667} 668 669uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 670{ 671 Mutex::Autolock _l(mLock); 672 PlaybackThread *thread = checkPlaybackThread_l(output); 673 if (thread == NULL) { 674 ALOGW("sampleRate() unknown thread %d", output); 675 return 0; 676 } 677 return thread->sampleRate(); 678} 679 680audio_format_t AudioFlinger::format(audio_io_handle_t output) const 681{ 682 Mutex::Autolock _l(mLock); 683 PlaybackThread *thread = checkPlaybackThread_l(output); 684 if (thread == NULL) { 685 ALOGW("format() unknown thread %d", output); 686 return AUDIO_FORMAT_INVALID; 687 } 688 return thread->format(); 689} 690 691size_t AudioFlinger::frameCount(audio_io_handle_t output) const 692{ 693 Mutex::Autolock _l(mLock); 694 PlaybackThread *thread = checkPlaybackThread_l(output); 695 if (thread == NULL) { 696 ALOGW("frameCount() unknown thread %d", output); 697 return 0; 698 } 699 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 700 // should examine all callers and fix them to handle smaller counts 701 return thread->frameCount(); 702} 703 704uint32_t AudioFlinger::latency(audio_io_handle_t output) const 705{ 706 Mutex::Autolock _l(mLock); 707 PlaybackThread *thread = checkPlaybackThread_l(output); 708 if (thread == NULL) { 709 ALOGW("latency(): no playback thread found for output handle %d", output); 710 return 0; 711 } 712 return thread->latency(); 713} 714 715status_t AudioFlinger::setMasterVolume(float value) 716{ 717 status_t ret = initCheck(); 718 if (ret != NO_ERROR) { 719 return ret; 720 } 721 722 // check calling permissions 723 if (!settingsAllowed()) { 724 return PERMISSION_DENIED; 725 } 726 727 Mutex::Autolock _l(mLock); 728 mMasterVolume = value; 729 730 // Set master volume in the HALs which support it. 731 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 732 AutoMutex lock(mHardwareLock); 733 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 734 735 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 736 if (dev->canSetMasterVolume()) { 737 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 738 } 739 mHardwareStatus = AUDIO_HW_IDLE; 740 } 741 742 // Now set the master volume in each playback thread. Playback threads 743 // assigned to HALs which do not have master volume support will apply 744 // master volume during the mix operation. Threads with HALs which do 745 // support master volume will simply ignore the setting. 746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 747 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 748 749 return NO_ERROR; 750} 751 752status_t AudioFlinger::setMode(audio_mode_t mode) 753{ 754 status_t ret = initCheck(); 755 if (ret != NO_ERROR) { 756 return ret; 757 } 758 759 // check calling permissions 760 if (!settingsAllowed()) { 761 return PERMISSION_DENIED; 762 } 763 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 764 ALOGW("Illegal value: setMode(%d)", mode); 765 return BAD_VALUE; 766 } 767 768 { // scope for the lock 769 AutoMutex lock(mHardwareLock); 770 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 771 mHardwareStatus = AUDIO_HW_SET_MODE; 772 ret = dev->set_mode(dev, mode); 773 mHardwareStatus = AUDIO_HW_IDLE; 774 } 775 776 if (NO_ERROR == ret) { 777 Mutex::Autolock _l(mLock); 778 mMode = mode; 779 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 780 mPlaybackThreads.valueAt(i)->setMode(mode); 781 } 782 783 return ret; 784} 785 786status_t AudioFlinger::setMicMute(bool state) 787{ 788 status_t ret = initCheck(); 789 if (ret != NO_ERROR) { 790 return ret; 791 } 792 793 // check calling permissions 794 if (!settingsAllowed()) { 795 return PERMISSION_DENIED; 796 } 797 798 AutoMutex lock(mHardwareLock); 799 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 800 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 801 ret = dev->set_mic_mute(dev, state); 802 mHardwareStatus = AUDIO_HW_IDLE; 803 return ret; 804} 805 806bool AudioFlinger::getMicMute() const 807{ 808 status_t ret = initCheck(); 809 if (ret != NO_ERROR) { 810 return false; 811 } 812 813 bool state = AUDIO_MODE_INVALID; 814 AutoMutex lock(mHardwareLock); 815 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 816 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 817 dev->get_mic_mute(dev, &state); 818 mHardwareStatus = AUDIO_HW_IDLE; 819 return state; 820} 821 822status_t AudioFlinger::setMasterMute(bool muted) 823{ 824 status_t ret = initCheck(); 825 if (ret != NO_ERROR) { 826 return ret; 827 } 828 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 Mutex::Autolock _l(mLock); 835 mMasterMute = muted; 836 837 // Set master mute in the HALs which support it. 838 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 839 AutoMutex lock(mHardwareLock); 840 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 841 842 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 843 if (dev->canSetMasterMute()) { 844 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 845 } 846 mHardwareStatus = AUDIO_HW_IDLE; 847 } 848 849 // Now set the master mute in each playback thread. Playback threads 850 // assigned to HALs which do not have master mute support will apply master 851 // mute during the mix operation. Threads with HALs which do support master 852 // mute will simply ignore the setting. 853 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 854 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 855 856 return NO_ERROR; 857} 858 859float AudioFlinger::masterVolume() const 860{ 861 Mutex::Autolock _l(mLock); 862 return masterVolume_l(); 863} 864 865bool AudioFlinger::masterMute() const 866{ 867 Mutex::Autolock _l(mLock); 868 return masterMute_l(); 869} 870 871float AudioFlinger::masterVolume_l() const 872{ 873 return mMasterVolume; 874} 875 876bool AudioFlinger::masterMute_l() const 877{ 878 return mMasterMute; 879} 880 881status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 882 audio_io_handle_t output) 883{ 884 // check calling permissions 885 if (!settingsAllowed()) { 886 return PERMISSION_DENIED; 887 } 888 889 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 890 ALOGE("setStreamVolume() invalid stream %d", stream); 891 return BAD_VALUE; 892 } 893 894 AutoMutex lock(mLock); 895 PlaybackThread *thread = NULL; 896 if (output != AUDIO_IO_HANDLE_NONE) { 897 thread = checkPlaybackThread_l(output); 898 if (thread == NULL) { 899 return BAD_VALUE; 900 } 901 } 902 903 mStreamTypes[stream].volume = value; 904 905 if (thread == NULL) { 906 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 907 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 908 } 909 } else { 910 thread->setStreamVolume(stream, value); 911 } 912 913 return NO_ERROR; 914} 915 916status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 917{ 918 // check calling permissions 919 if (!settingsAllowed()) { 920 return PERMISSION_DENIED; 921 } 922 923 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 924 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 925 ALOGE("setStreamMute() invalid stream %d", stream); 926 return BAD_VALUE; 927 } 928 929 AutoMutex lock(mLock); 930 mStreamTypes[stream].mute = muted; 931 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 932 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 933 934 return NO_ERROR; 935} 936 937float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 938{ 939 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 940 return 0.0f; 941 } 942 943 AutoMutex lock(mLock); 944 float volume; 945 if (output != AUDIO_IO_HANDLE_NONE) { 946 PlaybackThread *thread = checkPlaybackThread_l(output); 947 if (thread == NULL) { 948 return 0.0f; 949 } 950 volume = thread->streamVolume(stream); 951 } else { 952 volume = streamVolume_l(stream); 953 } 954 955 return volume; 956} 957 958bool AudioFlinger::streamMute(audio_stream_type_t stream) const 959{ 960 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 961 return true; 962 } 963 964 AutoMutex lock(mLock); 965 return streamMute_l(stream); 966} 967 968status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 969{ 970 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 971 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 972 973 // check calling permissions 974 if (!settingsAllowed()) { 975 return PERMISSION_DENIED; 976 } 977 978 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 979 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 980 Mutex::Autolock _l(mLock); 981 status_t final_result = NO_ERROR; 982 { 983 AutoMutex lock(mHardwareLock); 984 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 985 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 986 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 987 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 988 final_result = result ?: final_result; 989 } 990 mHardwareStatus = AUDIO_HW_IDLE; 991 } 992 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 993 AudioParameter param = AudioParameter(keyValuePairs); 994 String8 value; 995 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 996 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 997 if (mBtNrecIsOff != btNrecIsOff) { 998 for (size_t i = 0; i < mRecordThreads.size(); i++) { 999 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1000 audio_devices_t device = thread->inDevice(); 1001 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1002 // collect all of the thread's session IDs 1003 KeyedVector<int, bool> ids = thread->sessionIds(); 1004 // suspend effects associated with those session IDs 1005 for (size_t j = 0; j < ids.size(); ++j) { 1006 int sessionId = ids.keyAt(j); 1007 thread->setEffectSuspended(FX_IID_AEC, 1008 suspend, 1009 sessionId); 1010 thread->setEffectSuspended(FX_IID_NS, 1011 suspend, 1012 sessionId); 1013 } 1014 } 1015 mBtNrecIsOff = btNrecIsOff; 1016 } 1017 } 1018 String8 screenState; 1019 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1020 bool isOff = screenState == "off"; 1021 if (isOff != (AudioFlinger::mScreenState & 1)) { 1022 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1023 } 1024 } 1025 return final_result; 1026 } 1027 1028 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1029 // and the thread is exited once the lock is released 1030 sp<ThreadBase> thread; 1031 { 1032 Mutex::Autolock _l(mLock); 1033 thread = checkPlaybackThread_l(ioHandle); 1034 if (thread == 0) { 1035 thread = checkRecordThread_l(ioHandle); 1036 } else if (thread == primaryPlaybackThread_l()) { 1037 // indicate output device change to all input threads for pre processing 1038 AudioParameter param = AudioParameter(keyValuePairs); 1039 int value; 1040 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1041 (value != 0)) { 1042 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1043 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1044 } 1045 } 1046 } 1047 } 1048 if (thread != 0) { 1049 return thread->setParameters(keyValuePairs); 1050 } 1051 return BAD_VALUE; 1052} 1053 1054String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1055{ 1056 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1057 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1058 1059 Mutex::Autolock _l(mLock); 1060 1061 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1062 String8 out_s8; 1063 1064 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1065 char *s; 1066 { 1067 AutoMutex lock(mHardwareLock); 1068 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1069 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1070 s = dev->get_parameters(dev, keys.string()); 1071 mHardwareStatus = AUDIO_HW_IDLE; 1072 } 1073 out_s8 += String8(s ? s : ""); 1074 free(s); 1075 } 1076 return out_s8; 1077 } 1078 1079 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1080 if (playbackThread != NULL) { 1081 return playbackThread->getParameters(keys); 1082 } 1083 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1084 if (recordThread != NULL) { 1085 return recordThread->getParameters(keys); 1086 } 1087 return String8(""); 1088} 1089 1090size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1091 audio_channel_mask_t channelMask) const 1092{ 1093 status_t ret = initCheck(); 1094 if (ret != NO_ERROR) { 1095 return 0; 1096 } 1097 1098 AutoMutex lock(mHardwareLock); 1099 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1100 audio_config_t config; 1101 memset(&config, 0, sizeof(config)); 1102 config.sample_rate = sampleRate; 1103 config.channel_mask = channelMask; 1104 config.format = format; 1105 1106 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1107 size_t size = dev->get_input_buffer_size(dev, &config); 1108 mHardwareStatus = AUDIO_HW_IDLE; 1109 return size; 1110} 1111 1112uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1113{ 1114 Mutex::Autolock _l(mLock); 1115 1116 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1117 if (recordThread != NULL) { 1118 return recordThread->getInputFramesLost(); 1119 } 1120 return 0; 1121} 1122 1123status_t AudioFlinger::setVoiceVolume(float value) 1124{ 1125 status_t ret = initCheck(); 1126 if (ret != NO_ERROR) { 1127 return ret; 1128 } 1129 1130 // check calling permissions 1131 if (!settingsAllowed()) { 1132 return PERMISSION_DENIED; 1133 } 1134 1135 AutoMutex lock(mHardwareLock); 1136 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1137 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1138 ret = dev->set_voice_volume(dev, value); 1139 mHardwareStatus = AUDIO_HW_IDLE; 1140 1141 return ret; 1142} 1143 1144status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1145 audio_io_handle_t output) const 1146{ 1147 status_t status; 1148 1149 Mutex::Autolock _l(mLock); 1150 1151 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1152 if (playbackThread != NULL) { 1153 return playbackThread->getRenderPosition(halFrames, dspFrames); 1154 } 1155 1156 return BAD_VALUE; 1157} 1158 1159void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1160{ 1161 Mutex::Autolock _l(mLock); 1162 if (client == 0) { 1163 return; 1164 } 1165 bool clientAdded = false; 1166 { 1167 Mutex::Autolock _cl(mClientLock); 1168 1169 pid_t pid = IPCThreadState::self()->getCallingPid(); 1170 if (mNotificationClients.indexOfKey(pid) < 0) { 1171 sp<NotificationClient> notificationClient = new NotificationClient(this, 1172 client, 1173 pid); 1174 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1175 1176 mNotificationClients.add(pid, notificationClient); 1177 1178 sp<IBinder> binder = client->asBinder(); 1179 binder->linkToDeath(notificationClient); 1180 clientAdded = true; 1181 } 1182 } 1183 1184 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1185 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1186 if (clientAdded) { 1187 // the config change is always sent from playback or record threads to avoid deadlock 1188 // with AudioSystem::gLock 1189 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1190 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1191 } 1192 1193 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1194 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1195 } 1196 } 1197} 1198 1199void AudioFlinger::removeNotificationClient(pid_t pid) 1200{ 1201 Mutex::Autolock _l(mLock); 1202 { 1203 Mutex::Autolock _cl(mClientLock); 1204 mNotificationClients.removeItem(pid); 1205 } 1206 1207 ALOGV("%d died, releasing its sessions", pid); 1208 size_t num = mAudioSessionRefs.size(); 1209 bool removed = false; 1210 for (size_t i = 0; i< num; ) { 1211 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1212 ALOGV(" pid %d @ %d", ref->mPid, i); 1213 if (ref->mPid == pid) { 1214 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1215 mAudioSessionRefs.removeAt(i); 1216 delete ref; 1217 removed = true; 1218 num--; 1219 } else { 1220 i++; 1221 } 1222 } 1223 if (removed) { 1224 purgeStaleEffects_l(); 1225 } 1226} 1227 1228void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1229{ 1230 Mutex::Autolock _l(mClientLock); 1231 size_t size = mNotificationClients.size(); 1232 for (size_t i = 0; i < size; i++) { 1233 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1234 ioHandle, 1235 param2); 1236 } 1237} 1238 1239// removeClient_l() must be called with AudioFlinger::mClientLock held 1240void AudioFlinger::removeClient_l(pid_t pid) 1241{ 1242 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1243 IPCThreadState::self()->getCallingPid()); 1244 mClients.removeItem(pid); 1245} 1246 1247// getEffectThread_l() must be called with AudioFlinger::mLock held 1248sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1249{ 1250 sp<PlaybackThread> thread; 1251 1252 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1253 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1254 ALOG_ASSERT(thread == 0); 1255 thread = mPlaybackThreads.valueAt(i); 1256 } 1257 } 1258 1259 return thread; 1260} 1261 1262 1263 1264// ---------------------------------------------------------------------------- 1265 1266AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1267 : RefBase(), 1268 mAudioFlinger(audioFlinger), 1269 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1270 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1271 mPid(pid), 1272 mTimedTrackCount(0) 1273{ 1274 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1275} 1276 1277// Client destructor must be called with AudioFlinger::mClientLock held 1278AudioFlinger::Client::~Client() 1279{ 1280 mAudioFlinger->removeClient_l(mPid); 1281} 1282 1283sp<MemoryDealer> AudioFlinger::Client::heap() const 1284{ 1285 return mMemoryDealer; 1286} 1287 1288// Reserve one of the limited slots for a timed audio track associated 1289// with this client 1290bool AudioFlinger::Client::reserveTimedTrack() 1291{ 1292 const int kMaxTimedTracksPerClient = 4; 1293 1294 Mutex::Autolock _l(mTimedTrackLock); 1295 1296 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1297 ALOGW("can not create timed track - pid %d has exceeded the limit", 1298 mPid); 1299 return false; 1300 } 1301 1302 mTimedTrackCount++; 1303 return true; 1304} 1305 1306// Release a slot for a timed audio track 1307void AudioFlinger::Client::releaseTimedTrack() 1308{ 1309 Mutex::Autolock _l(mTimedTrackLock); 1310 mTimedTrackCount--; 1311} 1312 1313// ---------------------------------------------------------------------------- 1314 1315AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1316 const sp<IAudioFlingerClient>& client, 1317 pid_t pid) 1318 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1319{ 1320} 1321 1322AudioFlinger::NotificationClient::~NotificationClient() 1323{ 1324} 1325 1326void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1327{ 1328 sp<NotificationClient> keep(this); 1329 mAudioFlinger->removeNotificationClient(mPid); 1330} 1331 1332 1333// ---------------------------------------------------------------------------- 1334 1335static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1336 return audio_is_remote_submix_device(inDevice); 1337} 1338 1339sp<IAudioRecord> AudioFlinger::openRecord( 1340 audio_io_handle_t input, 1341 uint32_t sampleRate, 1342 audio_format_t format, 1343 audio_channel_mask_t channelMask, 1344 size_t *frameCount, 1345 IAudioFlinger::track_flags_t *flags, 1346 pid_t tid, 1347 int *sessionId, 1348 size_t *notificationFrames, 1349 sp<IMemory>& cblk, 1350 sp<IMemory>& buffers, 1351 status_t *status) 1352{ 1353 sp<RecordThread::RecordTrack> recordTrack; 1354 sp<RecordHandle> recordHandle; 1355 sp<Client> client; 1356 status_t lStatus; 1357 int lSessionId; 1358 1359 cblk.clear(); 1360 buffers.clear(); 1361 1362 // check calling permissions 1363 if (!recordingAllowed()) { 1364 ALOGE("openRecord() permission denied: recording not allowed"); 1365 lStatus = PERMISSION_DENIED; 1366 goto Exit; 1367 } 1368 1369 // further sample rate checks are performed by createRecordTrack_l() 1370 if (sampleRate == 0) { 1371 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 1376 // we don't yet support anything other than 16-bit PCM 1377 if (!(audio_is_valid_format(format) && 1378 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1379 ALOGE("openRecord() invalid format %#x", format); 1380 lStatus = BAD_VALUE; 1381 goto Exit; 1382 } 1383 1384 // further channel mask checks are performed by createRecordTrack_l() 1385 if (!audio_is_input_channel(channelMask)) { 1386 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1387 lStatus = BAD_VALUE; 1388 goto Exit; 1389 } 1390 1391 { 1392 Mutex::Autolock _l(mLock); 1393 RecordThread *thread = checkRecordThread_l(input); 1394 if (thread == NULL) { 1395 ALOGE("openRecord() checkRecordThread_l failed"); 1396 lStatus = BAD_VALUE; 1397 goto Exit; 1398 } 1399 1400 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1401 && !captureAudioOutputAllowed()) { 1402 ALOGE("openRecord() permission denied: capture not allowed"); 1403 lStatus = PERMISSION_DENIED; 1404 goto Exit; 1405 } 1406 1407 pid_t pid = IPCThreadState::self()->getCallingPid(); 1408 client = registerPid(pid); 1409 1410 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1411 lSessionId = *sessionId; 1412 } else { 1413 // if no audio session id is provided, create one here 1414 lSessionId = nextUniqueId(); 1415 if (sessionId != NULL) { 1416 *sessionId = lSessionId; 1417 } 1418 } 1419 ALOGV("openRecord() lSessionId: %d", lSessionId); 1420 1421 // TODO: the uid should be passed in as a parameter to openRecord 1422 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1423 frameCount, lSessionId, notificationFrames, 1424 IPCThreadState::self()->getCallingUid(), 1425 flags, tid, &lStatus); 1426 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1427 } 1428 1429 if (lStatus != NO_ERROR) { 1430 // remove local strong reference to Client before deleting the RecordTrack so that the 1431 // Client destructor is called by the TrackBase destructor with mClientLock held 1432 // Don't hold mClientLock when releasing the reference on the track as the 1433 // destructor will acquire it. 1434 { 1435 Mutex::Autolock _cl(mClientLock); 1436 client.clear(); 1437 } 1438 recordTrack.clear(); 1439 goto Exit; 1440 } 1441 1442 cblk = recordTrack->getCblk(); 1443 buffers = recordTrack->getBuffers(); 1444 1445 // return handle to client 1446 recordHandle = new RecordHandle(recordTrack); 1447 1448Exit: 1449 *status = lStatus; 1450 return recordHandle; 1451} 1452 1453 1454 1455// ---------------------------------------------------------------------------- 1456 1457audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1458{ 1459 if (name == NULL) { 1460 return 0; 1461 } 1462 if (!settingsAllowed()) { 1463 return 0; 1464 } 1465 Mutex::Autolock _l(mLock); 1466 return loadHwModule_l(name); 1467} 1468 1469// loadHwModule_l() must be called with AudioFlinger::mLock held 1470audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1471{ 1472 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1473 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1474 ALOGW("loadHwModule() module %s already loaded", name); 1475 return mAudioHwDevs.keyAt(i); 1476 } 1477 } 1478 1479 audio_hw_device_t *dev; 1480 1481 int rc = load_audio_interface(name, &dev); 1482 if (rc) { 1483 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1484 return 0; 1485 } 1486 1487 mHardwareStatus = AUDIO_HW_INIT; 1488 rc = dev->init_check(dev); 1489 mHardwareStatus = AUDIO_HW_IDLE; 1490 if (rc) { 1491 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1492 return 0; 1493 } 1494 1495 // Check and cache this HAL's level of support for master mute and master 1496 // volume. If this is the first HAL opened, and it supports the get 1497 // methods, use the initial values provided by the HAL as the current 1498 // master mute and volume settings. 1499 1500 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1501 { // scope for auto-lock pattern 1502 AutoMutex lock(mHardwareLock); 1503 1504 if (0 == mAudioHwDevs.size()) { 1505 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1506 if (NULL != dev->get_master_volume) { 1507 float mv; 1508 if (OK == dev->get_master_volume(dev, &mv)) { 1509 mMasterVolume = mv; 1510 } 1511 } 1512 1513 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1514 if (NULL != dev->get_master_mute) { 1515 bool mm; 1516 if (OK == dev->get_master_mute(dev, &mm)) { 1517 mMasterMute = mm; 1518 } 1519 } 1520 } 1521 1522 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1523 if ((NULL != dev->set_master_volume) && 1524 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1525 flags = static_cast<AudioHwDevice::Flags>(flags | 1526 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1527 } 1528 1529 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1530 if ((NULL != dev->set_master_mute) && 1531 (OK == dev->set_master_mute(dev, mMasterMute))) { 1532 flags = static_cast<AudioHwDevice::Flags>(flags | 1533 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1534 } 1535 1536 mHardwareStatus = AUDIO_HW_IDLE; 1537 } 1538 1539 audio_module_handle_t handle = nextUniqueId(); 1540 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1541 1542 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1543 name, dev->common.module->name, dev->common.module->id, handle); 1544 1545 return handle; 1546 1547} 1548 1549// ---------------------------------------------------------------------------- 1550 1551uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1552{ 1553 Mutex::Autolock _l(mLock); 1554 PlaybackThread *thread = primaryPlaybackThread_l(); 1555 return thread != NULL ? thread->sampleRate() : 0; 1556} 1557 1558size_t AudioFlinger::getPrimaryOutputFrameCount() 1559{ 1560 Mutex::Autolock _l(mLock); 1561 PlaybackThread *thread = primaryPlaybackThread_l(); 1562 return thread != NULL ? thread->frameCountHAL() : 0; 1563} 1564 1565// ---------------------------------------------------------------------------- 1566 1567status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1568{ 1569 uid_t uid = IPCThreadState::self()->getCallingUid(); 1570 if (uid != AID_SYSTEM) { 1571 return PERMISSION_DENIED; 1572 } 1573 Mutex::Autolock _l(mLock); 1574 if (mIsDeviceTypeKnown) { 1575 return INVALID_OPERATION; 1576 } 1577 mIsLowRamDevice = isLowRamDevice; 1578 mIsDeviceTypeKnown = true; 1579 return NO_ERROR; 1580} 1581 1582audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1583{ 1584 Mutex::Autolock _l(mLock); 1585 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1586 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1587 if ((thread->hasAudioSession(sessionId) & ThreadBase::TRACK_SESSION) != 0) { 1588 // A session can only be on one thread, so exit after first match 1589 String8 reply = thread->getParameters(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC)); 1590 AudioParameter param = AudioParameter(reply); 1591 int value; 1592 if (param.getInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value) == NO_ERROR) { 1593 return value; 1594 } 1595 break; 1596 } 1597 } 1598 return AUDIO_HW_SYNC_INVALID; 1599} 1600 1601// ---------------------------------------------------------------------------- 1602 1603 1604sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1605 audio_io_handle_t *output, 1606 audio_config_t *config, 1607 audio_devices_t devices, 1608 const String8& address, 1609 audio_output_flags_t flags) 1610{ 1611 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1612 if (outHwDev == NULL) { 1613 return 0; 1614 } 1615 1616 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1617 if (*output == AUDIO_IO_HANDLE_NONE) { 1618 *output = nextUniqueId(); 1619 } 1620 1621 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1622 1623 audio_stream_out_t *outStream = NULL; 1624 1625 // FOR TESTING ONLY: 1626 // This if statement allows overriding the audio policy settings 1627 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1628 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1629 // Check only for Normal Mixing mode 1630 if (kEnableExtendedPrecision) { 1631 // Specify format (uncomment one below to choose) 1632 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1633 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1634 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1635 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1636 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1637 } 1638 if (kEnableExtendedChannels) { 1639 // Specify channel mask (uncomment one below to choose) 1640 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1641 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1642 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1643 } 1644 } 1645 1646 status_t status = hwDevHal->open_output_stream(hwDevHal, 1647 *output, 1648 devices, 1649 flags, 1650 config, 1651 &outStream, 1652 address.string()); 1653 1654 mHardwareStatus = AUDIO_HW_IDLE; 1655 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1656 "channelMask %#x, status %d", 1657 outStream, 1658 config->sample_rate, 1659 config->format, 1660 config->channel_mask, 1661 status); 1662 1663 if (status == NO_ERROR && outStream != NULL) { 1664 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1665 1666 PlaybackThread *thread; 1667 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1668 thread = new OffloadThread(this, outputStream, *output, devices); 1669 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1670 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1671 || !isValidPcmSinkFormat(config->format) 1672 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1673 thread = new DirectOutputThread(this, outputStream, *output, devices); 1674 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1675 } else { 1676 thread = new MixerThread(this, outputStream, *output, devices); 1677 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1678 } 1679 mPlaybackThreads.add(*output, thread); 1680 return thread; 1681 } 1682 1683 return 0; 1684} 1685 1686status_t AudioFlinger::openOutput(audio_module_handle_t module, 1687 audio_io_handle_t *output, 1688 audio_config_t *config, 1689 audio_devices_t *devices, 1690 const String8& address, 1691 uint32_t *latencyMs, 1692 audio_output_flags_t flags) 1693{ 1694 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1695 module, 1696 (devices != NULL) ? *devices : 0, 1697 config->sample_rate, 1698 config->format, 1699 config->channel_mask, 1700 flags); 1701 1702 if (*devices == AUDIO_DEVICE_NONE) { 1703 return BAD_VALUE; 1704 } 1705 1706 Mutex::Autolock _l(mLock); 1707 1708 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1709 if (thread != 0) { 1710 *latencyMs = thread->latency(); 1711 1712 // notify client processes of the new output creation 1713 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1714 1715 // the first primary output opened designates the primary hw device 1716 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1717 ALOGI("Using module %d has the primary audio interface", module); 1718 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1719 1720 AutoMutex lock(mHardwareLock); 1721 mHardwareStatus = AUDIO_HW_SET_MODE; 1722 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1723 mHardwareStatus = AUDIO_HW_IDLE; 1724 1725 mPrimaryOutputSampleRate = config->sample_rate; 1726 } 1727 return NO_ERROR; 1728 } 1729 1730 return NO_INIT; 1731} 1732 1733audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1734 audio_io_handle_t output2) 1735{ 1736 Mutex::Autolock _l(mLock); 1737 MixerThread *thread1 = checkMixerThread_l(output1); 1738 MixerThread *thread2 = checkMixerThread_l(output2); 1739 1740 if (thread1 == NULL || thread2 == NULL) { 1741 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1742 output2); 1743 return AUDIO_IO_HANDLE_NONE; 1744 } 1745 1746 audio_io_handle_t id = nextUniqueId(); 1747 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1748 thread->addOutputTrack(thread2); 1749 mPlaybackThreads.add(id, thread); 1750 // notify client processes of the new output creation 1751 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1752 return id; 1753} 1754 1755status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1756{ 1757 return closeOutput_nonvirtual(output); 1758} 1759 1760status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1761{ 1762 // keep strong reference on the playback thread so that 1763 // it is not destroyed while exit() is executed 1764 sp<PlaybackThread> thread; 1765 { 1766 Mutex::Autolock _l(mLock); 1767 thread = checkPlaybackThread_l(output); 1768 if (thread == NULL) { 1769 return BAD_VALUE; 1770 } 1771 1772 ALOGV("closeOutput() %d", output); 1773 1774 if (thread->type() == ThreadBase::MIXER) { 1775 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1776 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1777 DuplicatingThread *dupThread = 1778 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1779 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1780 1781 } 1782 } 1783 } 1784 1785 1786 mPlaybackThreads.removeItem(output); 1787 // save all effects to the default thread 1788 if (mPlaybackThreads.size()) { 1789 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1790 if (dstThread != NULL) { 1791 // audioflinger lock is held here so the acquisition order of thread locks does not 1792 // matter 1793 Mutex::Autolock _dl(dstThread->mLock); 1794 Mutex::Autolock _sl(thread->mLock); 1795 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1796 for (size_t i = 0; i < effectChains.size(); i ++) { 1797 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1798 } 1799 } 1800 } 1801 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1802 } 1803 thread->exit(); 1804 // The thread entity (active unit of execution) is no longer running here, 1805 // but the ThreadBase container still exists. 1806 1807 if (thread->type() != ThreadBase::DUPLICATING) { 1808 closeOutputFinish(thread); 1809 } 1810 1811 return NO_ERROR; 1812} 1813 1814void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1815{ 1816 AudioStreamOut *out = thread->clearOutput(); 1817 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1818 // from now on thread->mOutput is NULL 1819 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1820 delete out; 1821} 1822 1823void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1824{ 1825 mPlaybackThreads.removeItem(thread->mId); 1826 thread->exit(); 1827 closeOutputFinish(thread); 1828} 1829 1830status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1831{ 1832 Mutex::Autolock _l(mLock); 1833 PlaybackThread *thread = checkPlaybackThread_l(output); 1834 1835 if (thread == NULL) { 1836 return BAD_VALUE; 1837 } 1838 1839 ALOGV("suspendOutput() %d", output); 1840 thread->suspend(); 1841 1842 return NO_ERROR; 1843} 1844 1845status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 PlaybackThread *thread = checkPlaybackThread_l(output); 1849 1850 if (thread == NULL) { 1851 return BAD_VALUE; 1852 } 1853 1854 ALOGV("restoreOutput() %d", output); 1855 1856 thread->restore(); 1857 1858 return NO_ERROR; 1859} 1860 1861status_t AudioFlinger::openInput(audio_module_handle_t module, 1862 audio_io_handle_t *input, 1863 audio_config_t *config, 1864 audio_devices_t *device, 1865 const String8& address, 1866 audio_source_t source, 1867 audio_input_flags_t flags) 1868{ 1869 Mutex::Autolock _l(mLock); 1870 1871 if (*device == AUDIO_DEVICE_NONE) { 1872 return BAD_VALUE; 1873 } 1874 1875 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1876 1877 if (thread != 0) { 1878 // notify client processes of the new input creation 1879 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1880 return NO_ERROR; 1881 } 1882 return NO_INIT; 1883} 1884 1885sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1886 audio_io_handle_t *input, 1887 audio_config_t *config, 1888 audio_devices_t device, 1889 const String8& address, 1890 audio_source_t source, 1891 audio_input_flags_t flags) 1892{ 1893 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1894 if (inHwDev == NULL) { 1895 *input = AUDIO_IO_HANDLE_NONE; 1896 return 0; 1897 } 1898 1899 if (*input == AUDIO_IO_HANDLE_NONE) { 1900 *input = nextUniqueId(); 1901 } 1902 1903 audio_config_t halconfig = *config; 1904 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1905 audio_stream_in_t *inStream = NULL; 1906 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1907 &inStream, flags, address.string(), source); 1908 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1909 ", Format %#x, Channels %x, flags %#x, status %d", 1910 inStream, 1911 halconfig.sample_rate, 1912 halconfig.format, 1913 halconfig.channel_mask, 1914 flags, 1915 status); 1916 1917 // If the input could not be opened with the requested parameters and we can handle the 1918 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1919 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1920 if (status == BAD_VALUE && 1921 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1922 (halconfig.sample_rate <= 2 * config->sample_rate) && 1923 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1924 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1925 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1926 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 1927 inStream = NULL; 1928 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1929 &inStream, flags, address.string(), source); 1930 // FIXME log this new status; HAL should not propose any further changes 1931 } 1932 1933 if (status == NO_ERROR && inStream != NULL) { 1934 1935#ifdef TEE_SINK 1936 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1937 // or (re-)create if current Pipe is idle and does not match the new format 1938 sp<NBAIO_Sink> teeSink; 1939 enum { 1940 TEE_SINK_NO, // don't copy input 1941 TEE_SINK_NEW, // copy input using a new pipe 1942 TEE_SINK_OLD, // copy input using an existing pipe 1943 } kind; 1944 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 1945 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 1946 if (!mTeeSinkInputEnabled) { 1947 kind = TEE_SINK_NO; 1948 } else if (!Format_isValid(format)) { 1949 kind = TEE_SINK_NO; 1950 } else if (mRecordTeeSink == 0) { 1951 kind = TEE_SINK_NEW; 1952 } else if (mRecordTeeSink->getStrongCount() != 1) { 1953 kind = TEE_SINK_NO; 1954 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1955 kind = TEE_SINK_OLD; 1956 } else { 1957 kind = TEE_SINK_NEW; 1958 } 1959 switch (kind) { 1960 case TEE_SINK_NEW: { 1961 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1962 size_t numCounterOffers = 0; 1963 const NBAIO_Format offers[1] = {format}; 1964 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1965 ALOG_ASSERT(index == 0); 1966 PipeReader *pipeReader = new PipeReader(*pipe); 1967 numCounterOffers = 0; 1968 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1969 ALOG_ASSERT(index == 0); 1970 mRecordTeeSink = pipe; 1971 mRecordTeeSource = pipeReader; 1972 teeSink = pipe; 1973 } 1974 break; 1975 case TEE_SINK_OLD: 1976 teeSink = mRecordTeeSink; 1977 break; 1978 case TEE_SINK_NO: 1979 default: 1980 break; 1981 } 1982#endif 1983 1984 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 1985 1986 // Start record thread 1987 // RecordThread requires both input and output device indication to forward to audio 1988 // pre processing modules 1989 sp<RecordThread> thread = new RecordThread(this, 1990 inputStream, 1991 *input, 1992 primaryOutputDevice_l(), 1993 device 1994#ifdef TEE_SINK 1995 , teeSink 1996#endif 1997 ); 1998 mRecordThreads.add(*input, thread); 1999 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2000 return thread; 2001 } 2002 2003 *input = AUDIO_IO_HANDLE_NONE; 2004 return 0; 2005} 2006 2007status_t AudioFlinger::closeInput(audio_io_handle_t input) 2008{ 2009 return closeInput_nonvirtual(input); 2010} 2011 2012status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2013{ 2014 // keep strong reference on the record thread so that 2015 // it is not destroyed while exit() is executed 2016 sp<RecordThread> thread; 2017 { 2018 Mutex::Autolock _l(mLock); 2019 thread = checkRecordThread_l(input); 2020 if (thread == 0) { 2021 return BAD_VALUE; 2022 } 2023 2024 ALOGV("closeInput() %d", input); 2025 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2026 mRecordThreads.removeItem(input); 2027 } 2028 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2029 // we have a different lock for notification client 2030 closeInputFinish(thread); 2031 return NO_ERROR; 2032} 2033 2034void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2035{ 2036 thread->exit(); 2037 AudioStreamIn *in = thread->clearInput(); 2038 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2039 // from now on thread->mInput is NULL 2040 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2041 delete in; 2042} 2043 2044void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2045{ 2046 mRecordThreads.removeItem(thread->mId); 2047 closeInputFinish(thread); 2048} 2049 2050status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2051{ 2052 Mutex::Autolock _l(mLock); 2053 ALOGV("invalidateStream() stream %d", stream); 2054 2055 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2056 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2057 thread->invalidateTracks(stream); 2058 } 2059 2060 return NO_ERROR; 2061} 2062 2063 2064audio_unique_id_t AudioFlinger::newAudioUniqueId() 2065{ 2066 return nextUniqueId(); 2067} 2068 2069void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2070{ 2071 Mutex::Autolock _l(mLock); 2072 pid_t caller = IPCThreadState::self()->getCallingPid(); 2073 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2074 if (pid != -1 && (caller == getpid_cached)) { 2075 caller = pid; 2076 } 2077 2078 { 2079 Mutex::Autolock _cl(mClientLock); 2080 // Ignore requests received from processes not known as notification client. The request 2081 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2082 // called from a different pid leaving a stale session reference. Also we don't know how 2083 // to clear this reference if the client process dies. 2084 if (mNotificationClients.indexOfKey(caller) < 0) { 2085 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2086 return; 2087 } 2088 } 2089 2090 size_t num = mAudioSessionRefs.size(); 2091 for (size_t i = 0; i< num; i++) { 2092 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2093 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2094 ref->mCnt++; 2095 ALOGV(" incremented refcount to %d", ref->mCnt); 2096 return; 2097 } 2098 } 2099 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2100 ALOGV(" added new entry for %d", audioSession); 2101} 2102 2103void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2104{ 2105 Mutex::Autolock _l(mLock); 2106 pid_t caller = IPCThreadState::self()->getCallingPid(); 2107 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2108 if (pid != -1 && (caller == getpid_cached)) { 2109 caller = pid; 2110 } 2111 size_t num = mAudioSessionRefs.size(); 2112 for (size_t i = 0; i< num; i++) { 2113 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2114 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2115 ref->mCnt--; 2116 ALOGV(" decremented refcount to %d", ref->mCnt); 2117 if (ref->mCnt == 0) { 2118 mAudioSessionRefs.removeAt(i); 2119 delete ref; 2120 purgeStaleEffects_l(); 2121 } 2122 return; 2123 } 2124 } 2125 // If the caller is mediaserver it is likely that the session being released was acquired 2126 // on behalf of a process not in notification clients and we ignore the warning. 2127 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2128} 2129 2130void AudioFlinger::purgeStaleEffects_l() { 2131 2132 ALOGV("purging stale effects"); 2133 2134 Vector< sp<EffectChain> > chains; 2135 2136 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2137 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2138 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2139 sp<EffectChain> ec = t->mEffectChains[j]; 2140 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2141 chains.push(ec); 2142 } 2143 } 2144 } 2145 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2146 sp<RecordThread> t = mRecordThreads.valueAt(i); 2147 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2148 sp<EffectChain> ec = t->mEffectChains[j]; 2149 chains.push(ec); 2150 } 2151 } 2152 2153 for (size_t i = 0; i < chains.size(); i++) { 2154 sp<EffectChain> ec = chains[i]; 2155 int sessionid = ec->sessionId(); 2156 sp<ThreadBase> t = ec->mThread.promote(); 2157 if (t == 0) { 2158 continue; 2159 } 2160 size_t numsessionrefs = mAudioSessionRefs.size(); 2161 bool found = false; 2162 for (size_t k = 0; k < numsessionrefs; k++) { 2163 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2164 if (ref->mSessionid == sessionid) { 2165 ALOGV(" session %d still exists for %d with %d refs", 2166 sessionid, ref->mPid, ref->mCnt); 2167 found = true; 2168 break; 2169 } 2170 } 2171 if (!found) { 2172 Mutex::Autolock _l(t->mLock); 2173 // remove all effects from the chain 2174 while (ec->mEffects.size()) { 2175 sp<EffectModule> effect = ec->mEffects[0]; 2176 effect->unPin(); 2177 t->removeEffect_l(effect); 2178 if (effect->purgeHandles()) { 2179 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2180 } 2181 AudioSystem::unregisterEffect(effect->id()); 2182 } 2183 } 2184 } 2185 return; 2186} 2187 2188// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2189AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2190{ 2191 return mPlaybackThreads.valueFor(output).get(); 2192} 2193 2194// checkMixerThread_l() must be called with AudioFlinger::mLock held 2195AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2196{ 2197 PlaybackThread *thread = checkPlaybackThread_l(output); 2198 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2199} 2200 2201// checkRecordThread_l() must be called with AudioFlinger::mLock held 2202AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2203{ 2204 return mRecordThreads.valueFor(input).get(); 2205} 2206 2207uint32_t AudioFlinger::nextUniqueId() 2208{ 2209 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2210} 2211 2212AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2213{ 2214 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2215 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2216 AudioStreamOut *output = thread->getOutput(); 2217 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2218 return thread; 2219 } 2220 } 2221 return NULL; 2222} 2223 2224audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2225{ 2226 PlaybackThread *thread = primaryPlaybackThread_l(); 2227 2228 if (thread == NULL) { 2229 return 0; 2230 } 2231 2232 return thread->outDevice(); 2233} 2234 2235sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2236 int triggerSession, 2237 int listenerSession, 2238 sync_event_callback_t callBack, 2239 wp<RefBase> cookie) 2240{ 2241 Mutex::Autolock _l(mLock); 2242 2243 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2244 status_t playStatus = NAME_NOT_FOUND; 2245 status_t recStatus = NAME_NOT_FOUND; 2246 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2247 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2248 if (playStatus == NO_ERROR) { 2249 return event; 2250 } 2251 } 2252 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2253 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2254 if (recStatus == NO_ERROR) { 2255 return event; 2256 } 2257 } 2258 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2259 mPendingSyncEvents.add(event); 2260 } else { 2261 ALOGV("createSyncEvent() invalid event %d", event->type()); 2262 event.clear(); 2263 } 2264 return event; 2265} 2266 2267// ---------------------------------------------------------------------------- 2268// Effect management 2269// ---------------------------------------------------------------------------- 2270 2271 2272status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2273{ 2274 Mutex::Autolock _l(mLock); 2275 return EffectQueryNumberEffects(numEffects); 2276} 2277 2278status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2279{ 2280 Mutex::Autolock _l(mLock); 2281 return EffectQueryEffect(index, descriptor); 2282} 2283 2284status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2285 effect_descriptor_t *descriptor) const 2286{ 2287 Mutex::Autolock _l(mLock); 2288 return EffectGetDescriptor(pUuid, descriptor); 2289} 2290 2291 2292sp<IEffect> AudioFlinger::createEffect( 2293 effect_descriptor_t *pDesc, 2294 const sp<IEffectClient>& effectClient, 2295 int32_t priority, 2296 audio_io_handle_t io, 2297 int sessionId, 2298 status_t *status, 2299 int *id, 2300 int *enabled) 2301{ 2302 status_t lStatus = NO_ERROR; 2303 sp<EffectHandle> handle; 2304 effect_descriptor_t desc; 2305 2306 pid_t pid = IPCThreadState::self()->getCallingPid(); 2307 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2308 pid, effectClient.get(), priority, sessionId, io); 2309 2310 if (pDesc == NULL) { 2311 lStatus = BAD_VALUE; 2312 goto Exit; 2313 } 2314 2315 // check audio settings permission for global effects 2316 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2317 lStatus = PERMISSION_DENIED; 2318 goto Exit; 2319 } 2320 2321 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2322 // that can only be created by audio policy manager (running in same process) 2323 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2324 lStatus = PERMISSION_DENIED; 2325 goto Exit; 2326 } 2327 2328 { 2329 if (!EffectIsNullUuid(&pDesc->uuid)) { 2330 // if uuid is specified, request effect descriptor 2331 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2332 if (lStatus < 0) { 2333 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2334 goto Exit; 2335 } 2336 } else { 2337 // if uuid is not specified, look for an available implementation 2338 // of the required type in effect factory 2339 if (EffectIsNullUuid(&pDesc->type)) { 2340 ALOGW("createEffect() no effect type"); 2341 lStatus = BAD_VALUE; 2342 goto Exit; 2343 } 2344 uint32_t numEffects = 0; 2345 effect_descriptor_t d; 2346 d.flags = 0; // prevent compiler warning 2347 bool found = false; 2348 2349 lStatus = EffectQueryNumberEffects(&numEffects); 2350 if (lStatus < 0) { 2351 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2352 goto Exit; 2353 } 2354 for (uint32_t i = 0; i < numEffects; i++) { 2355 lStatus = EffectQueryEffect(i, &desc); 2356 if (lStatus < 0) { 2357 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2358 continue; 2359 } 2360 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2361 // If matching type found save effect descriptor. If the session is 2362 // 0 and the effect is not auxiliary, continue enumeration in case 2363 // an auxiliary version of this effect type is available 2364 found = true; 2365 d = desc; 2366 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2367 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2368 break; 2369 } 2370 } 2371 } 2372 if (!found) { 2373 lStatus = BAD_VALUE; 2374 ALOGW("createEffect() effect not found"); 2375 goto Exit; 2376 } 2377 // For same effect type, chose auxiliary version over insert version if 2378 // connect to output mix (Compliance to OpenSL ES) 2379 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2380 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2381 desc = d; 2382 } 2383 } 2384 2385 // Do not allow auxiliary effects on a session different from 0 (output mix) 2386 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2387 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2388 lStatus = INVALID_OPERATION; 2389 goto Exit; 2390 } 2391 2392 // check recording permission for visualizer 2393 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2394 !recordingAllowed()) { 2395 lStatus = PERMISSION_DENIED; 2396 goto Exit; 2397 } 2398 2399 // return effect descriptor 2400 *pDesc = desc; 2401 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2402 // if the output returned by getOutputForEffect() is removed before we lock the 2403 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2404 // and we will exit safely 2405 io = AudioSystem::getOutputForEffect(&desc); 2406 ALOGV("createEffect got output %d", io); 2407 } 2408 2409 Mutex::Autolock _l(mLock); 2410 2411 // If output is not specified try to find a matching audio session ID in one of the 2412 // output threads. 2413 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2414 // because of code checking output when entering the function. 2415 // Note: io is never 0 when creating an effect on an input 2416 if (io == AUDIO_IO_HANDLE_NONE) { 2417 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2418 // output must be specified by AudioPolicyManager when using session 2419 // AUDIO_SESSION_OUTPUT_STAGE 2420 lStatus = BAD_VALUE; 2421 goto Exit; 2422 } 2423 // look for the thread where the specified audio session is present 2424 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2425 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2426 io = mPlaybackThreads.keyAt(i); 2427 break; 2428 } 2429 } 2430 if (io == 0) { 2431 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2432 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2433 io = mRecordThreads.keyAt(i); 2434 break; 2435 } 2436 } 2437 } 2438 // If no output thread contains the requested session ID, default to 2439 // first output. The effect chain will be moved to the correct output 2440 // thread when a track with the same session ID is created 2441 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2442 io = mPlaybackThreads.keyAt(0); 2443 } 2444 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2445 } 2446 ThreadBase *thread = checkRecordThread_l(io); 2447 if (thread == NULL) { 2448 thread = checkPlaybackThread_l(io); 2449 if (thread == NULL) { 2450 ALOGE("createEffect() unknown output thread"); 2451 lStatus = BAD_VALUE; 2452 goto Exit; 2453 } 2454 } 2455 2456 sp<Client> client = registerPid(pid); 2457 2458 // create effect on selected output thread 2459 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2460 &desc, enabled, &lStatus); 2461 if (handle != 0 && id != NULL) { 2462 *id = handle->id(); 2463 } 2464 if (handle == 0) { 2465 // remove local strong reference to Client with mClientLock held 2466 Mutex::Autolock _cl(mClientLock); 2467 client.clear(); 2468 } 2469 } 2470 2471Exit: 2472 *status = lStatus; 2473 return handle; 2474} 2475 2476status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2477 audio_io_handle_t dstOutput) 2478{ 2479 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2480 sessionId, srcOutput, dstOutput); 2481 Mutex::Autolock _l(mLock); 2482 if (srcOutput == dstOutput) { 2483 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2484 return NO_ERROR; 2485 } 2486 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2487 if (srcThread == NULL) { 2488 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2489 return BAD_VALUE; 2490 } 2491 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2492 if (dstThread == NULL) { 2493 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2494 return BAD_VALUE; 2495 } 2496 2497 Mutex::Autolock _dl(dstThread->mLock); 2498 Mutex::Autolock _sl(srcThread->mLock); 2499 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2500} 2501 2502// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2503status_t AudioFlinger::moveEffectChain_l(int sessionId, 2504 AudioFlinger::PlaybackThread *srcThread, 2505 AudioFlinger::PlaybackThread *dstThread, 2506 bool reRegister) 2507{ 2508 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2509 sessionId, srcThread, dstThread); 2510 2511 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2512 if (chain == 0) { 2513 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2514 sessionId, srcThread); 2515 return INVALID_OPERATION; 2516 } 2517 2518 // Check whether the destination thread has a channel count of FCC_2, which is 2519 // currently required for (most) effects. Prevent moving the effect chain here rather 2520 // than disabling the addEffect_l() call in dstThread below. 2521 if (dstThread->mChannelCount != FCC_2) { 2522 ALOGW("moveEffectChain_l() effect chain failed because" 2523 " destination thread %p channel count(%u) != %u", 2524 dstThread, dstThread->mChannelCount, FCC_2); 2525 return INVALID_OPERATION; 2526 } 2527 2528 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2529 // so that a new chain is created with correct parameters when first effect is added. This is 2530 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2531 // removed. 2532 srcThread->removeEffectChain_l(chain); 2533 2534 // transfer all effects one by one so that new effect chain is created on new thread with 2535 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2536 sp<EffectChain> dstChain; 2537 uint32_t strategy = 0; // prevent compiler warning 2538 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2539 Vector< sp<EffectModule> > removed; 2540 status_t status = NO_ERROR; 2541 while (effect != 0) { 2542 srcThread->removeEffect_l(effect); 2543 removed.add(effect); 2544 status = dstThread->addEffect_l(effect); 2545 if (status != NO_ERROR) { 2546 break; 2547 } 2548 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2549 if (effect->state() == EffectModule::ACTIVE || 2550 effect->state() == EffectModule::STOPPING) { 2551 effect->start(); 2552 } 2553 // if the move request is not received from audio policy manager, the effect must be 2554 // re-registered with the new strategy and output 2555 if (dstChain == 0) { 2556 dstChain = effect->chain().promote(); 2557 if (dstChain == 0) { 2558 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2559 status = NO_INIT; 2560 break; 2561 } 2562 strategy = dstChain->strategy(); 2563 } 2564 if (reRegister) { 2565 AudioSystem::unregisterEffect(effect->id()); 2566 AudioSystem::registerEffect(&effect->desc(), 2567 dstThread->id(), 2568 strategy, 2569 sessionId, 2570 effect->id()); 2571 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2572 } 2573 effect = chain->getEffectFromId_l(0); 2574 } 2575 2576 if (status != NO_ERROR) { 2577 for (size_t i = 0; i < removed.size(); i++) { 2578 srcThread->addEffect_l(removed[i]); 2579 if (dstChain != 0 && reRegister) { 2580 AudioSystem::unregisterEffect(removed[i]->id()); 2581 AudioSystem::registerEffect(&removed[i]->desc(), 2582 srcThread->id(), 2583 strategy, 2584 sessionId, 2585 removed[i]->id()); 2586 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2587 } 2588 } 2589 } 2590 2591 return status; 2592} 2593 2594bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2595{ 2596 if (mGlobalEffectEnableTime != 0 && 2597 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2598 return true; 2599 } 2600 2601 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2602 sp<EffectChain> ec = 2603 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2604 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2605 return true; 2606 } 2607 } 2608 return false; 2609} 2610 2611void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2612{ 2613 Mutex::Autolock _l(mLock); 2614 2615 mGlobalEffectEnableTime = systemTime(); 2616 2617 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2618 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2619 if (t->mType == ThreadBase::OFFLOAD) { 2620 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2621 } 2622 } 2623 2624} 2625 2626struct Entry { 2627#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2628 char mName[MAX_NAME]; 2629}; 2630 2631int comparEntry(const void *p1, const void *p2) 2632{ 2633 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2634} 2635 2636#ifdef TEE_SINK 2637void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2638{ 2639 NBAIO_Source *teeSource = source.get(); 2640 if (teeSource != NULL) { 2641 // .wav rotation 2642 // There is a benign race condition if 2 threads call this simultaneously. 2643 // They would both traverse the directory, but the result would simply be 2644 // failures at unlink() which are ignored. It's also unlikely since 2645 // normally dumpsys is only done by bugreport or from the command line. 2646 char teePath[32+256]; 2647 strcpy(teePath, "/data/misc/media"); 2648 size_t teePathLen = strlen(teePath); 2649 DIR *dir = opendir(teePath); 2650 teePath[teePathLen++] = '/'; 2651 if (dir != NULL) { 2652#define MAX_SORT 20 // number of entries to sort 2653#define MAX_KEEP 10 // number of entries to keep 2654 struct Entry entries[MAX_SORT]; 2655 size_t entryCount = 0; 2656 while (entryCount < MAX_SORT) { 2657 struct dirent de; 2658 struct dirent *result = NULL; 2659 int rc = readdir_r(dir, &de, &result); 2660 if (rc != 0) { 2661 ALOGW("readdir_r failed %d", rc); 2662 break; 2663 } 2664 if (result == NULL) { 2665 break; 2666 } 2667 if (result != &de) { 2668 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2669 break; 2670 } 2671 // ignore non .wav file entries 2672 size_t nameLen = strlen(de.d_name); 2673 if (nameLen <= 4 || nameLen >= MAX_NAME || 2674 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2675 continue; 2676 } 2677 strcpy(entries[entryCount++].mName, de.d_name); 2678 } 2679 (void) closedir(dir); 2680 if (entryCount > MAX_KEEP) { 2681 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2682 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2683 strcpy(&teePath[teePathLen], entries[i].mName); 2684 (void) unlink(teePath); 2685 } 2686 } 2687 } else { 2688 if (fd >= 0) { 2689 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2690 } 2691 } 2692 char teeTime[16]; 2693 struct timeval tv; 2694 gettimeofday(&tv, NULL); 2695 struct tm tm; 2696 localtime_r(&tv.tv_sec, &tm); 2697 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2698 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2699 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2700 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2701 if (teeFd >= 0) { 2702 // FIXME use libsndfile 2703 char wavHeader[44]; 2704 memcpy(wavHeader, 2705 "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", 2706 sizeof(wavHeader)); 2707 NBAIO_Format format = teeSource->format(); 2708 unsigned channelCount = Format_channelCount(format); 2709 uint32_t sampleRate = Format_sampleRate(format); 2710 size_t frameSize = Format_frameSize(format); 2711 wavHeader[22] = channelCount; // number of channels 2712 wavHeader[24] = sampleRate; // sample rate 2713 wavHeader[25] = sampleRate >> 8; 2714 wavHeader[32] = frameSize; // block alignment 2715 wavHeader[33] = frameSize >> 8; 2716 write(teeFd, wavHeader, sizeof(wavHeader)); 2717 size_t total = 0; 2718 bool firstRead = true; 2719#define TEE_SINK_READ 1024 // frames per I/O operation 2720 void *buffer = malloc(TEE_SINK_READ * frameSize); 2721 for (;;) { 2722 size_t count = TEE_SINK_READ; 2723 ssize_t actual = teeSource->read(buffer, count, 2724 AudioBufferProvider::kInvalidPTS); 2725 bool wasFirstRead = firstRead; 2726 firstRead = false; 2727 if (actual <= 0) { 2728 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2729 continue; 2730 } 2731 break; 2732 } 2733 ALOG_ASSERT(actual <= (ssize_t)count); 2734 write(teeFd, buffer, actual * frameSize); 2735 total += actual; 2736 } 2737 free(buffer); 2738 lseek(teeFd, (off_t) 4, SEEK_SET); 2739 uint32_t temp = 44 + total * frameSize - 8; 2740 // FIXME not big-endian safe 2741 write(teeFd, &temp, sizeof(temp)); 2742 lseek(teeFd, (off_t) 40, SEEK_SET); 2743 temp = total * frameSize; 2744 // FIXME not big-endian safe 2745 write(teeFd, &temp, sizeof(temp)); 2746 close(teeFd); 2747 if (fd >= 0) { 2748 dprintf(fd, "tee copied to %s\n", teePath); 2749 } 2750 } else { 2751 if (fd >= 0) { 2752 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2753 } 2754 } 2755 } 2756} 2757#endif 2758 2759// ---------------------------------------------------------------------------- 2760 2761status_t AudioFlinger::onTransact( 2762 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2763{ 2764 return BnAudioFlinger::onTransact(code, data, reply, flags); 2765} 2766 2767}; // namespace android 2768