AudioFlinger.cpp revision 2f035f59d1e28728d38d18a7f0f7a9c6e8b0c11b
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 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 800 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 801 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 802 status_t result = dev->set_mic_mute(dev, state); 803 if (result != NO_ERROR) { 804 ret = result; 805 } 806 } 807 mHardwareStatus = AUDIO_HW_IDLE; 808 return ret; 809} 810 811bool AudioFlinger::getMicMute() const 812{ 813 status_t ret = initCheck(); 814 if (ret != NO_ERROR) { 815 return false; 816 } 817 818 bool state = AUDIO_MODE_INVALID; 819 AutoMutex lock(mHardwareLock); 820 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 821 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 822 dev->get_mic_mute(dev, &state); 823 mHardwareStatus = AUDIO_HW_IDLE; 824 return state; 825} 826 827status_t AudioFlinger::setMasterMute(bool muted) 828{ 829 status_t ret = initCheck(); 830 if (ret != NO_ERROR) { 831 return ret; 832 } 833 834 // check calling permissions 835 if (!settingsAllowed()) { 836 return PERMISSION_DENIED; 837 } 838 839 Mutex::Autolock _l(mLock); 840 mMasterMute = muted; 841 842 // Set master mute in the HALs which support it. 843 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 844 AutoMutex lock(mHardwareLock); 845 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 846 847 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 848 if (dev->canSetMasterMute()) { 849 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 850 } 851 mHardwareStatus = AUDIO_HW_IDLE; 852 } 853 854 // Now set the master mute in each playback thread. Playback threads 855 // assigned to HALs which do not have master mute support will apply master 856 // mute during the mix operation. Threads with HALs which do support master 857 // mute will simply ignore the setting. 858 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 859 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 860 861 return NO_ERROR; 862} 863 864float AudioFlinger::masterVolume() const 865{ 866 Mutex::Autolock _l(mLock); 867 return masterVolume_l(); 868} 869 870bool AudioFlinger::masterMute() const 871{ 872 Mutex::Autolock _l(mLock); 873 return masterMute_l(); 874} 875 876float AudioFlinger::masterVolume_l() const 877{ 878 return mMasterVolume; 879} 880 881bool AudioFlinger::masterMute_l() const 882{ 883 return mMasterMute; 884} 885 886status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 887 audio_io_handle_t output) 888{ 889 // check calling permissions 890 if (!settingsAllowed()) { 891 return PERMISSION_DENIED; 892 } 893 894 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 895 ALOGE("setStreamVolume() invalid stream %d", stream); 896 return BAD_VALUE; 897 } 898 899 AutoMutex lock(mLock); 900 PlaybackThread *thread = NULL; 901 if (output != AUDIO_IO_HANDLE_NONE) { 902 thread = checkPlaybackThread_l(output); 903 if (thread == NULL) { 904 return BAD_VALUE; 905 } 906 } 907 908 mStreamTypes[stream].volume = value; 909 910 if (thread == NULL) { 911 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 912 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 913 } 914 } else { 915 thread->setStreamVolume(stream, value); 916 } 917 918 return NO_ERROR; 919} 920 921status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 922{ 923 // check calling permissions 924 if (!settingsAllowed()) { 925 return PERMISSION_DENIED; 926 } 927 928 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 929 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 930 ALOGE("setStreamMute() invalid stream %d", stream); 931 return BAD_VALUE; 932 } 933 934 AutoMutex lock(mLock); 935 mStreamTypes[stream].mute = muted; 936 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 937 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 938 939 return NO_ERROR; 940} 941 942float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 943{ 944 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 945 return 0.0f; 946 } 947 948 AutoMutex lock(mLock); 949 float volume; 950 if (output != AUDIO_IO_HANDLE_NONE) { 951 PlaybackThread *thread = checkPlaybackThread_l(output); 952 if (thread == NULL) { 953 return 0.0f; 954 } 955 volume = thread->streamVolume(stream); 956 } else { 957 volume = streamVolume_l(stream); 958 } 959 960 return volume; 961} 962 963bool AudioFlinger::streamMute(audio_stream_type_t stream) const 964{ 965 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 966 return true; 967 } 968 969 AutoMutex lock(mLock); 970 return streamMute_l(stream); 971} 972 973status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 974{ 975 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 976 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 977 978 // check calling permissions 979 if (!settingsAllowed()) { 980 return PERMISSION_DENIED; 981 } 982 983 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 984 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 985 Mutex::Autolock _l(mLock); 986 status_t final_result = NO_ERROR; 987 { 988 AutoMutex lock(mHardwareLock); 989 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 990 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 991 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 992 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 993 final_result = result ?: final_result; 994 } 995 mHardwareStatus = AUDIO_HW_IDLE; 996 } 997 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 998 AudioParameter param = AudioParameter(keyValuePairs); 999 String8 value; 1000 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1001 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1002 if (mBtNrecIsOff != btNrecIsOff) { 1003 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1004 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1005 audio_devices_t device = thread->inDevice(); 1006 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1007 // collect all of the thread's session IDs 1008 KeyedVector<int, bool> ids = thread->sessionIds(); 1009 // suspend effects associated with those session IDs 1010 for (size_t j = 0; j < ids.size(); ++j) { 1011 int sessionId = ids.keyAt(j); 1012 thread->setEffectSuspended(FX_IID_AEC, 1013 suspend, 1014 sessionId); 1015 thread->setEffectSuspended(FX_IID_NS, 1016 suspend, 1017 sessionId); 1018 } 1019 } 1020 mBtNrecIsOff = btNrecIsOff; 1021 } 1022 } 1023 String8 screenState; 1024 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1025 bool isOff = screenState == "off"; 1026 if (isOff != (AudioFlinger::mScreenState & 1)) { 1027 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1028 } 1029 } 1030 return final_result; 1031 } 1032 1033 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1034 // and the thread is exited once the lock is released 1035 sp<ThreadBase> thread; 1036 { 1037 Mutex::Autolock _l(mLock); 1038 thread = checkPlaybackThread_l(ioHandle); 1039 if (thread == 0) { 1040 thread = checkRecordThread_l(ioHandle); 1041 } else if (thread == primaryPlaybackThread_l()) { 1042 // indicate output device change to all input threads for pre processing 1043 AudioParameter param = AudioParameter(keyValuePairs); 1044 int value; 1045 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1046 (value != 0)) { 1047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1048 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1049 } 1050 } 1051 } 1052 } 1053 if (thread != 0) { 1054 return thread->setParameters(keyValuePairs); 1055 } 1056 return BAD_VALUE; 1057} 1058 1059String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1060{ 1061 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1062 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1063 1064 Mutex::Autolock _l(mLock); 1065 1066 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1067 String8 out_s8; 1068 1069 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1070 char *s; 1071 { 1072 AutoMutex lock(mHardwareLock); 1073 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1074 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1075 s = dev->get_parameters(dev, keys.string()); 1076 mHardwareStatus = AUDIO_HW_IDLE; 1077 } 1078 out_s8 += String8(s ? s : ""); 1079 free(s); 1080 } 1081 return out_s8; 1082 } 1083 1084 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1085 if (playbackThread != NULL) { 1086 return playbackThread->getParameters(keys); 1087 } 1088 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1089 if (recordThread != NULL) { 1090 return recordThread->getParameters(keys); 1091 } 1092 return String8(""); 1093} 1094 1095size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1096 audio_channel_mask_t channelMask) const 1097{ 1098 status_t ret = initCheck(); 1099 if (ret != NO_ERROR) { 1100 return 0; 1101 } 1102 1103 AutoMutex lock(mHardwareLock); 1104 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1105 audio_config_t config; 1106 memset(&config, 0, sizeof(config)); 1107 config.sample_rate = sampleRate; 1108 config.channel_mask = channelMask; 1109 config.format = format; 1110 1111 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1112 size_t size = dev->get_input_buffer_size(dev, &config); 1113 mHardwareStatus = AUDIO_HW_IDLE; 1114 return size; 1115} 1116 1117uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1118{ 1119 Mutex::Autolock _l(mLock); 1120 1121 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1122 if (recordThread != NULL) { 1123 return recordThread->getInputFramesLost(); 1124 } 1125 return 0; 1126} 1127 1128status_t AudioFlinger::setVoiceVolume(float value) 1129{ 1130 status_t ret = initCheck(); 1131 if (ret != NO_ERROR) { 1132 return ret; 1133 } 1134 1135 // check calling permissions 1136 if (!settingsAllowed()) { 1137 return PERMISSION_DENIED; 1138 } 1139 1140 AutoMutex lock(mHardwareLock); 1141 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1142 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1143 ret = dev->set_voice_volume(dev, value); 1144 mHardwareStatus = AUDIO_HW_IDLE; 1145 1146 return ret; 1147} 1148 1149status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1150 audio_io_handle_t output) const 1151{ 1152 status_t status; 1153 1154 Mutex::Autolock _l(mLock); 1155 1156 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1157 if (playbackThread != NULL) { 1158 return playbackThread->getRenderPosition(halFrames, dspFrames); 1159 } 1160 1161 return BAD_VALUE; 1162} 1163 1164void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1165{ 1166 Mutex::Autolock _l(mLock); 1167 if (client == 0) { 1168 return; 1169 } 1170 bool clientAdded = false; 1171 { 1172 Mutex::Autolock _cl(mClientLock); 1173 1174 pid_t pid = IPCThreadState::self()->getCallingPid(); 1175 if (mNotificationClients.indexOfKey(pid) < 0) { 1176 sp<NotificationClient> notificationClient = new NotificationClient(this, 1177 client, 1178 pid); 1179 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1180 1181 mNotificationClients.add(pid, notificationClient); 1182 1183 sp<IBinder> binder = client->asBinder(); 1184 binder->linkToDeath(notificationClient); 1185 clientAdded = true; 1186 } 1187 } 1188 1189 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1190 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1191 if (clientAdded) { 1192 // the config change is always sent from playback or record threads to avoid deadlock 1193 // with AudioSystem::gLock 1194 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1195 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1196 } 1197 1198 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1199 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1200 } 1201 } 1202} 1203 1204void AudioFlinger::removeNotificationClient(pid_t pid) 1205{ 1206 Mutex::Autolock _l(mLock); 1207 { 1208 Mutex::Autolock _cl(mClientLock); 1209 mNotificationClients.removeItem(pid); 1210 } 1211 1212 ALOGV("%d died, releasing its sessions", pid); 1213 size_t num = mAudioSessionRefs.size(); 1214 bool removed = false; 1215 for (size_t i = 0; i< num; ) { 1216 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1217 ALOGV(" pid %d @ %d", ref->mPid, i); 1218 if (ref->mPid == pid) { 1219 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1220 mAudioSessionRefs.removeAt(i); 1221 delete ref; 1222 removed = true; 1223 num--; 1224 } else { 1225 i++; 1226 } 1227 } 1228 if (removed) { 1229 purgeStaleEffects_l(); 1230 } 1231} 1232 1233void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1234{ 1235 Mutex::Autolock _l(mClientLock); 1236 size_t size = mNotificationClients.size(); 1237 for (size_t i = 0; i < size; i++) { 1238 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1239 ioHandle, 1240 param2); 1241 } 1242} 1243 1244// removeClient_l() must be called with AudioFlinger::mClientLock held 1245void AudioFlinger::removeClient_l(pid_t pid) 1246{ 1247 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1248 IPCThreadState::self()->getCallingPid()); 1249 mClients.removeItem(pid); 1250} 1251 1252// getEffectThread_l() must be called with AudioFlinger::mLock held 1253sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1254{ 1255 sp<PlaybackThread> thread; 1256 1257 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1258 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1259 ALOG_ASSERT(thread == 0); 1260 thread = mPlaybackThreads.valueAt(i); 1261 } 1262 } 1263 1264 return thread; 1265} 1266 1267 1268 1269// ---------------------------------------------------------------------------- 1270 1271AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1272 : RefBase(), 1273 mAudioFlinger(audioFlinger), 1274 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1275 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1276 mPid(pid), 1277 mTimedTrackCount(0) 1278{ 1279 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1280} 1281 1282// Client destructor must be called with AudioFlinger::mClientLock held 1283AudioFlinger::Client::~Client() 1284{ 1285 mAudioFlinger->removeClient_l(mPid); 1286} 1287 1288sp<MemoryDealer> AudioFlinger::Client::heap() const 1289{ 1290 return mMemoryDealer; 1291} 1292 1293// Reserve one of the limited slots for a timed audio track associated 1294// with this client 1295bool AudioFlinger::Client::reserveTimedTrack() 1296{ 1297 const int kMaxTimedTracksPerClient = 4; 1298 1299 Mutex::Autolock _l(mTimedTrackLock); 1300 1301 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1302 ALOGW("can not create timed track - pid %d has exceeded the limit", 1303 mPid); 1304 return false; 1305 } 1306 1307 mTimedTrackCount++; 1308 return true; 1309} 1310 1311// Release a slot for a timed audio track 1312void AudioFlinger::Client::releaseTimedTrack() 1313{ 1314 Mutex::Autolock _l(mTimedTrackLock); 1315 mTimedTrackCount--; 1316} 1317 1318// ---------------------------------------------------------------------------- 1319 1320AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1321 const sp<IAudioFlingerClient>& client, 1322 pid_t pid) 1323 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1324{ 1325} 1326 1327AudioFlinger::NotificationClient::~NotificationClient() 1328{ 1329} 1330 1331void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1332{ 1333 sp<NotificationClient> keep(this); 1334 mAudioFlinger->removeNotificationClient(mPid); 1335} 1336 1337 1338// ---------------------------------------------------------------------------- 1339 1340static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1341 return audio_is_remote_submix_device(inDevice); 1342} 1343 1344sp<IAudioRecord> AudioFlinger::openRecord( 1345 audio_io_handle_t input, 1346 uint32_t sampleRate, 1347 audio_format_t format, 1348 audio_channel_mask_t channelMask, 1349 size_t *frameCount, 1350 IAudioFlinger::track_flags_t *flags, 1351 pid_t tid, 1352 int *sessionId, 1353 size_t *notificationFrames, 1354 sp<IMemory>& cblk, 1355 sp<IMemory>& buffers, 1356 status_t *status) 1357{ 1358 sp<RecordThread::RecordTrack> recordTrack; 1359 sp<RecordHandle> recordHandle; 1360 sp<Client> client; 1361 status_t lStatus; 1362 int lSessionId; 1363 1364 cblk.clear(); 1365 buffers.clear(); 1366 1367 // check calling permissions 1368 if (!recordingAllowed()) { 1369 ALOGE("openRecord() permission denied: recording not allowed"); 1370 lStatus = PERMISSION_DENIED; 1371 goto Exit; 1372 } 1373 1374 // further sample rate checks are performed by createRecordTrack_l() 1375 if (sampleRate == 0) { 1376 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1377 lStatus = BAD_VALUE; 1378 goto Exit; 1379 } 1380 1381 // we don't yet support anything other than 16-bit PCM 1382 if (!(audio_is_valid_format(format) && 1383 audio_is_linear_pcm(format) && format == AUDIO_FORMAT_PCM_16_BIT)) { 1384 ALOGE("openRecord() invalid format %#x", format); 1385 lStatus = BAD_VALUE; 1386 goto Exit; 1387 } 1388 1389 // further channel mask checks are performed by createRecordTrack_l() 1390 if (!audio_is_input_channel(channelMask)) { 1391 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1392 lStatus = BAD_VALUE; 1393 goto Exit; 1394 } 1395 1396 { 1397 Mutex::Autolock _l(mLock); 1398 RecordThread *thread = checkRecordThread_l(input); 1399 if (thread == NULL) { 1400 ALOGE("openRecord() checkRecordThread_l failed"); 1401 lStatus = BAD_VALUE; 1402 goto Exit; 1403 } 1404 1405 if (deviceRequiresCaptureAudioOutputPermission(thread->inDevice()) 1406 && !captureAudioOutputAllowed()) { 1407 ALOGE("openRecord() permission denied: capture not allowed"); 1408 lStatus = PERMISSION_DENIED; 1409 goto Exit; 1410 } 1411 1412 pid_t pid = IPCThreadState::self()->getCallingPid(); 1413 client = registerPid(pid); 1414 1415 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1416 lSessionId = *sessionId; 1417 } else { 1418 // if no audio session id is provided, create one here 1419 lSessionId = nextUniqueId(); 1420 if (sessionId != NULL) { 1421 *sessionId = lSessionId; 1422 } 1423 } 1424 ALOGV("openRecord() lSessionId: %d", lSessionId); 1425 1426 // TODO: the uid should be passed in as a parameter to openRecord 1427 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1428 frameCount, lSessionId, notificationFrames, 1429 IPCThreadState::self()->getCallingUid(), 1430 flags, tid, &lStatus); 1431 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1432 } 1433 1434 if (lStatus != NO_ERROR) { 1435 // remove local strong reference to Client before deleting the RecordTrack so that the 1436 // Client destructor is called by the TrackBase destructor with mClientLock held 1437 // Don't hold mClientLock when releasing the reference on the track as the 1438 // destructor will acquire it. 1439 { 1440 Mutex::Autolock _cl(mClientLock); 1441 client.clear(); 1442 } 1443 recordTrack.clear(); 1444 goto Exit; 1445 } 1446 1447 cblk = recordTrack->getCblk(); 1448 buffers = recordTrack->getBuffers(); 1449 1450 // return handle to client 1451 recordHandle = new RecordHandle(recordTrack); 1452 1453Exit: 1454 *status = lStatus; 1455 return recordHandle; 1456} 1457 1458 1459 1460// ---------------------------------------------------------------------------- 1461 1462audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1463{ 1464 if (name == NULL) { 1465 return 0; 1466 } 1467 if (!settingsAllowed()) { 1468 return 0; 1469 } 1470 Mutex::Autolock _l(mLock); 1471 return loadHwModule_l(name); 1472} 1473 1474// loadHwModule_l() must be called with AudioFlinger::mLock held 1475audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1476{ 1477 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1478 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1479 ALOGW("loadHwModule() module %s already loaded", name); 1480 return mAudioHwDevs.keyAt(i); 1481 } 1482 } 1483 1484 audio_hw_device_t *dev; 1485 1486 int rc = load_audio_interface(name, &dev); 1487 if (rc) { 1488 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1489 return 0; 1490 } 1491 1492 mHardwareStatus = AUDIO_HW_INIT; 1493 rc = dev->init_check(dev); 1494 mHardwareStatus = AUDIO_HW_IDLE; 1495 if (rc) { 1496 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1497 return 0; 1498 } 1499 1500 // Check and cache this HAL's level of support for master mute and master 1501 // volume. If this is the first HAL opened, and it supports the get 1502 // methods, use the initial values provided by the HAL as the current 1503 // master mute and volume settings. 1504 1505 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1506 { // scope for auto-lock pattern 1507 AutoMutex lock(mHardwareLock); 1508 1509 if (0 == mAudioHwDevs.size()) { 1510 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1511 if (NULL != dev->get_master_volume) { 1512 float mv; 1513 if (OK == dev->get_master_volume(dev, &mv)) { 1514 mMasterVolume = mv; 1515 } 1516 } 1517 1518 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1519 if (NULL != dev->get_master_mute) { 1520 bool mm; 1521 if (OK == dev->get_master_mute(dev, &mm)) { 1522 mMasterMute = mm; 1523 } 1524 } 1525 } 1526 1527 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1528 if ((NULL != dev->set_master_volume) && 1529 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1530 flags = static_cast<AudioHwDevice::Flags>(flags | 1531 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1532 } 1533 1534 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1535 if ((NULL != dev->set_master_mute) && 1536 (OK == dev->set_master_mute(dev, mMasterMute))) { 1537 flags = static_cast<AudioHwDevice::Flags>(flags | 1538 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1539 } 1540 1541 mHardwareStatus = AUDIO_HW_IDLE; 1542 } 1543 1544 audio_module_handle_t handle = nextUniqueId(); 1545 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1546 1547 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1548 name, dev->common.module->name, dev->common.module->id, handle); 1549 1550 return handle; 1551 1552} 1553 1554// ---------------------------------------------------------------------------- 1555 1556uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1557{ 1558 Mutex::Autolock _l(mLock); 1559 PlaybackThread *thread = primaryPlaybackThread_l(); 1560 return thread != NULL ? thread->sampleRate() : 0; 1561} 1562 1563size_t AudioFlinger::getPrimaryOutputFrameCount() 1564{ 1565 Mutex::Autolock _l(mLock); 1566 PlaybackThread *thread = primaryPlaybackThread_l(); 1567 return thread != NULL ? thread->frameCountHAL() : 0; 1568} 1569 1570// ---------------------------------------------------------------------------- 1571 1572status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1573{ 1574 uid_t uid = IPCThreadState::self()->getCallingUid(); 1575 if (uid != AID_SYSTEM) { 1576 return PERMISSION_DENIED; 1577 } 1578 Mutex::Autolock _l(mLock); 1579 if (mIsDeviceTypeKnown) { 1580 return INVALID_OPERATION; 1581 } 1582 mIsLowRamDevice = isLowRamDevice; 1583 mIsDeviceTypeKnown = true; 1584 return NO_ERROR; 1585} 1586 1587audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1588{ 1589 Mutex::Autolock _l(mLock); 1590 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1591 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1592 if ((thread->hasAudioSession(sessionId) & ThreadBase::TRACK_SESSION) != 0) { 1593 // A session can only be on one thread, so exit after first match 1594 String8 reply = thread->getParameters(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC)); 1595 AudioParameter param = AudioParameter(reply); 1596 int value; 1597 if (param.getInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value) == NO_ERROR) { 1598 return value; 1599 } 1600 break; 1601 } 1602 } 1603 return AUDIO_HW_SYNC_INVALID; 1604} 1605 1606// ---------------------------------------------------------------------------- 1607 1608 1609sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1610 audio_io_handle_t *output, 1611 audio_config_t *config, 1612 audio_devices_t devices, 1613 const String8& address, 1614 audio_output_flags_t flags) 1615{ 1616 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1617 if (outHwDev == NULL) { 1618 return 0; 1619 } 1620 1621 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1622 if (*output == AUDIO_IO_HANDLE_NONE) { 1623 *output = nextUniqueId(); 1624 } 1625 1626 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1627 1628 audio_stream_out_t *outStream = NULL; 1629 1630 // FOR TESTING ONLY: 1631 // This if statement allows overriding the audio policy settings 1632 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1633 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1634 // Check only for Normal Mixing mode 1635 if (kEnableExtendedPrecision) { 1636 // Specify format (uncomment one below to choose) 1637 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1638 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1639 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1640 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1641 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1642 } 1643 if (kEnableExtendedChannels) { 1644 // Specify channel mask (uncomment one below to choose) 1645 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1646 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1647 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1648 } 1649 } 1650 1651 status_t status = hwDevHal->open_output_stream(hwDevHal, 1652 *output, 1653 devices, 1654 flags, 1655 config, 1656 &outStream, 1657 address.string()); 1658 1659 mHardwareStatus = AUDIO_HW_IDLE; 1660 ALOGV("openOutput_l() openOutputStream returned output %p, sampleRate %d, Format %#x, " 1661 "channelMask %#x, status %d", 1662 outStream, 1663 config->sample_rate, 1664 config->format, 1665 config->channel_mask, 1666 status); 1667 1668 if (status == NO_ERROR && outStream != NULL) { 1669 AudioStreamOut *outputStream = new AudioStreamOut(outHwDev, outStream, flags); 1670 1671 PlaybackThread *thread; 1672 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1673 thread = new OffloadThread(this, outputStream, *output, devices); 1674 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1675 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1676 || !isValidPcmSinkFormat(config->format) 1677 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1678 thread = new DirectOutputThread(this, outputStream, *output, devices); 1679 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1680 } else { 1681 thread = new MixerThread(this, outputStream, *output, devices); 1682 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1683 } 1684 mPlaybackThreads.add(*output, thread); 1685 return thread; 1686 } 1687 1688 return 0; 1689} 1690 1691status_t AudioFlinger::openOutput(audio_module_handle_t module, 1692 audio_io_handle_t *output, 1693 audio_config_t *config, 1694 audio_devices_t *devices, 1695 const String8& address, 1696 uint32_t *latencyMs, 1697 audio_output_flags_t flags) 1698{ 1699 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1700 module, 1701 (devices != NULL) ? *devices : 0, 1702 config->sample_rate, 1703 config->format, 1704 config->channel_mask, 1705 flags); 1706 1707 if (*devices == AUDIO_DEVICE_NONE) { 1708 return BAD_VALUE; 1709 } 1710 1711 Mutex::Autolock _l(mLock); 1712 1713 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1714 if (thread != 0) { 1715 *latencyMs = thread->latency(); 1716 1717 // notify client processes of the new output creation 1718 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1719 1720 // the first primary output opened designates the primary hw device 1721 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1722 ALOGI("Using module %d has the primary audio interface", module); 1723 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1724 1725 AutoMutex lock(mHardwareLock); 1726 mHardwareStatus = AUDIO_HW_SET_MODE; 1727 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1728 mHardwareStatus = AUDIO_HW_IDLE; 1729 1730 mPrimaryOutputSampleRate = config->sample_rate; 1731 } 1732 return NO_ERROR; 1733 } 1734 1735 return NO_INIT; 1736} 1737 1738audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1739 audio_io_handle_t output2) 1740{ 1741 Mutex::Autolock _l(mLock); 1742 MixerThread *thread1 = checkMixerThread_l(output1); 1743 MixerThread *thread2 = checkMixerThread_l(output2); 1744 1745 if (thread1 == NULL || thread2 == NULL) { 1746 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1747 output2); 1748 return AUDIO_IO_HANDLE_NONE; 1749 } 1750 1751 audio_io_handle_t id = nextUniqueId(); 1752 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1753 thread->addOutputTrack(thread2); 1754 mPlaybackThreads.add(id, thread); 1755 // notify client processes of the new output creation 1756 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1757 return id; 1758} 1759 1760status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1761{ 1762 return closeOutput_nonvirtual(output); 1763} 1764 1765status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1766{ 1767 // keep strong reference on the playback thread so that 1768 // it is not destroyed while exit() is executed 1769 sp<PlaybackThread> thread; 1770 { 1771 Mutex::Autolock _l(mLock); 1772 thread = checkPlaybackThread_l(output); 1773 if (thread == NULL) { 1774 return BAD_VALUE; 1775 } 1776 1777 ALOGV("closeOutput() %d", output); 1778 1779 if (thread->type() == ThreadBase::MIXER) { 1780 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1781 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1782 DuplicatingThread *dupThread = 1783 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1784 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1785 1786 } 1787 } 1788 } 1789 1790 1791 mPlaybackThreads.removeItem(output); 1792 // save all effects to the default thread 1793 if (mPlaybackThreads.size()) { 1794 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1795 if (dstThread != NULL) { 1796 // audioflinger lock is held here so the acquisition order of thread locks does not 1797 // matter 1798 Mutex::Autolock _dl(dstThread->mLock); 1799 Mutex::Autolock _sl(thread->mLock); 1800 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1801 for (size_t i = 0; i < effectChains.size(); i ++) { 1802 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1803 } 1804 } 1805 } 1806 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1807 } 1808 thread->exit(); 1809 // The thread entity (active unit of execution) is no longer running here, 1810 // but the ThreadBase container still exists. 1811 1812 if (thread->type() != ThreadBase::DUPLICATING) { 1813 closeOutputFinish(thread); 1814 } 1815 1816 return NO_ERROR; 1817} 1818 1819void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1820{ 1821 AudioStreamOut *out = thread->clearOutput(); 1822 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1823 // from now on thread->mOutput is NULL 1824 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1825 delete out; 1826} 1827 1828void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1829{ 1830 mPlaybackThreads.removeItem(thread->mId); 1831 thread->exit(); 1832 closeOutputFinish(thread); 1833} 1834 1835status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 PlaybackThread *thread = checkPlaybackThread_l(output); 1839 1840 if (thread == NULL) { 1841 return BAD_VALUE; 1842 } 1843 1844 ALOGV("suspendOutput() %d", output); 1845 thread->suspend(); 1846 1847 return NO_ERROR; 1848} 1849 1850status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1851{ 1852 Mutex::Autolock _l(mLock); 1853 PlaybackThread *thread = checkPlaybackThread_l(output); 1854 1855 if (thread == NULL) { 1856 return BAD_VALUE; 1857 } 1858 1859 ALOGV("restoreOutput() %d", output); 1860 1861 thread->restore(); 1862 1863 return NO_ERROR; 1864} 1865 1866status_t AudioFlinger::openInput(audio_module_handle_t module, 1867 audio_io_handle_t *input, 1868 audio_config_t *config, 1869 audio_devices_t *device, 1870 const String8& address, 1871 audio_source_t source, 1872 audio_input_flags_t flags) 1873{ 1874 Mutex::Autolock _l(mLock); 1875 1876 if (*device == AUDIO_DEVICE_NONE) { 1877 return BAD_VALUE; 1878 } 1879 1880 sp<RecordThread> thread = openInput_l(module, input, config, *device, address, source, flags); 1881 1882 if (thread != 0) { 1883 // notify client processes of the new input creation 1884 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1885 return NO_ERROR; 1886 } 1887 return NO_INIT; 1888} 1889 1890sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 1891 audio_io_handle_t *input, 1892 audio_config_t *config, 1893 audio_devices_t device, 1894 const String8& address, 1895 audio_source_t source, 1896 audio_input_flags_t flags) 1897{ 1898 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, device); 1899 if (inHwDev == NULL) { 1900 *input = AUDIO_IO_HANDLE_NONE; 1901 return 0; 1902 } 1903 1904 if (*input == AUDIO_IO_HANDLE_NONE) { 1905 *input = nextUniqueId(); 1906 } 1907 1908 audio_config_t halconfig = *config; 1909 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 1910 audio_stream_in_t *inStream = NULL; 1911 status_t status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1912 &inStream, flags, address.string(), source); 1913 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 1914 ", Format %#x, Channels %x, flags %#x, status %d", 1915 inStream, 1916 halconfig.sample_rate, 1917 halconfig.format, 1918 halconfig.channel_mask, 1919 flags, 1920 status); 1921 1922 // If the input could not be opened with the requested parameters and we can handle the 1923 // conversion internally, try to open again with the proposed parameters. The AudioFlinger can 1924 // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs. 1925 if (status == BAD_VALUE && 1926 config->format == halconfig.format && halconfig.format == AUDIO_FORMAT_PCM_16_BIT && 1927 (halconfig.sample_rate <= 2 * config->sample_rate) && 1928 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 1929 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 1930 // FIXME describe the change proposed by HAL (save old values so we can log them here) 1931 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 1932 inStream = NULL; 1933 status = inHwHal->open_input_stream(inHwHal, *input, device, &halconfig, 1934 &inStream, flags, address.string(), source); 1935 // FIXME log this new status; HAL should not propose any further changes 1936 } 1937 1938 if (status == NO_ERROR && inStream != NULL) { 1939 1940#ifdef TEE_SINK 1941 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 1942 // or (re-)create if current Pipe is idle and does not match the new format 1943 sp<NBAIO_Sink> teeSink; 1944 enum { 1945 TEE_SINK_NO, // don't copy input 1946 TEE_SINK_NEW, // copy input using a new pipe 1947 TEE_SINK_OLD, // copy input using an existing pipe 1948 } kind; 1949 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 1950 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 1951 if (!mTeeSinkInputEnabled) { 1952 kind = TEE_SINK_NO; 1953 } else if (!Format_isValid(format)) { 1954 kind = TEE_SINK_NO; 1955 } else if (mRecordTeeSink == 0) { 1956 kind = TEE_SINK_NEW; 1957 } else if (mRecordTeeSink->getStrongCount() != 1) { 1958 kind = TEE_SINK_NO; 1959 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 1960 kind = TEE_SINK_OLD; 1961 } else { 1962 kind = TEE_SINK_NEW; 1963 } 1964 switch (kind) { 1965 case TEE_SINK_NEW: { 1966 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 1967 size_t numCounterOffers = 0; 1968 const NBAIO_Format offers[1] = {format}; 1969 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 1970 ALOG_ASSERT(index == 0); 1971 PipeReader *pipeReader = new PipeReader(*pipe); 1972 numCounterOffers = 0; 1973 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 1974 ALOG_ASSERT(index == 0); 1975 mRecordTeeSink = pipe; 1976 mRecordTeeSource = pipeReader; 1977 teeSink = pipe; 1978 } 1979 break; 1980 case TEE_SINK_OLD: 1981 teeSink = mRecordTeeSink; 1982 break; 1983 case TEE_SINK_NO: 1984 default: 1985 break; 1986 } 1987#endif 1988 1989 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 1990 1991 // Start record thread 1992 // RecordThread requires both input and output device indication to forward to audio 1993 // pre processing modules 1994 sp<RecordThread> thread = new RecordThread(this, 1995 inputStream, 1996 *input, 1997 primaryOutputDevice_l(), 1998 device 1999#ifdef TEE_SINK 2000 , teeSink 2001#endif 2002 ); 2003 mRecordThreads.add(*input, thread); 2004 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2005 return thread; 2006 } 2007 2008 *input = AUDIO_IO_HANDLE_NONE; 2009 return 0; 2010} 2011 2012status_t AudioFlinger::closeInput(audio_io_handle_t input) 2013{ 2014 return closeInput_nonvirtual(input); 2015} 2016 2017status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2018{ 2019 // keep strong reference on the record thread so that 2020 // it is not destroyed while exit() is executed 2021 sp<RecordThread> thread; 2022 { 2023 Mutex::Autolock _l(mLock); 2024 thread = checkRecordThread_l(input); 2025 if (thread == 0) { 2026 return BAD_VALUE; 2027 } 2028 2029 ALOGV("closeInput() %d", input); 2030 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2031 mRecordThreads.removeItem(input); 2032 } 2033 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2034 // we have a different lock for notification client 2035 closeInputFinish(thread); 2036 return NO_ERROR; 2037} 2038 2039void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2040{ 2041 thread->exit(); 2042 AudioStreamIn *in = thread->clearInput(); 2043 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2044 // from now on thread->mInput is NULL 2045 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2046 delete in; 2047} 2048 2049void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2050{ 2051 mRecordThreads.removeItem(thread->mId); 2052 closeInputFinish(thread); 2053} 2054 2055status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2056{ 2057 Mutex::Autolock _l(mLock); 2058 ALOGV("invalidateStream() stream %d", stream); 2059 2060 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2061 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2062 thread->invalidateTracks(stream); 2063 } 2064 2065 return NO_ERROR; 2066} 2067 2068 2069audio_unique_id_t AudioFlinger::newAudioUniqueId() 2070{ 2071 return nextUniqueId(); 2072} 2073 2074void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2075{ 2076 Mutex::Autolock _l(mLock); 2077 pid_t caller = IPCThreadState::self()->getCallingPid(); 2078 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2079 if (pid != -1 && (caller == getpid_cached)) { 2080 caller = pid; 2081 } 2082 2083 { 2084 Mutex::Autolock _cl(mClientLock); 2085 // Ignore requests received from processes not known as notification client. The request 2086 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2087 // called from a different pid leaving a stale session reference. Also we don't know how 2088 // to clear this reference if the client process dies. 2089 if (mNotificationClients.indexOfKey(caller) < 0) { 2090 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2091 return; 2092 } 2093 } 2094 2095 size_t num = mAudioSessionRefs.size(); 2096 for (size_t i = 0; i< num; i++) { 2097 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2098 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2099 ref->mCnt++; 2100 ALOGV(" incremented refcount to %d", ref->mCnt); 2101 return; 2102 } 2103 } 2104 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2105 ALOGV(" added new entry for %d", audioSession); 2106} 2107 2108void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2109{ 2110 Mutex::Autolock _l(mLock); 2111 pid_t caller = IPCThreadState::self()->getCallingPid(); 2112 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2113 if (pid != -1 && (caller == getpid_cached)) { 2114 caller = pid; 2115 } 2116 size_t num = mAudioSessionRefs.size(); 2117 for (size_t i = 0; i< num; i++) { 2118 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2119 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2120 ref->mCnt--; 2121 ALOGV(" decremented refcount to %d", ref->mCnt); 2122 if (ref->mCnt == 0) { 2123 mAudioSessionRefs.removeAt(i); 2124 delete ref; 2125 purgeStaleEffects_l(); 2126 } 2127 return; 2128 } 2129 } 2130 // If the caller is mediaserver it is likely that the session being released was acquired 2131 // on behalf of a process not in notification clients and we ignore the warning. 2132 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2133} 2134 2135void AudioFlinger::purgeStaleEffects_l() { 2136 2137 ALOGV("purging stale effects"); 2138 2139 Vector< sp<EffectChain> > chains; 2140 2141 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2142 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2143 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2144 sp<EffectChain> ec = t->mEffectChains[j]; 2145 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2146 chains.push(ec); 2147 } 2148 } 2149 } 2150 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2151 sp<RecordThread> t = mRecordThreads.valueAt(i); 2152 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2153 sp<EffectChain> ec = t->mEffectChains[j]; 2154 chains.push(ec); 2155 } 2156 } 2157 2158 for (size_t i = 0; i < chains.size(); i++) { 2159 sp<EffectChain> ec = chains[i]; 2160 int sessionid = ec->sessionId(); 2161 sp<ThreadBase> t = ec->mThread.promote(); 2162 if (t == 0) { 2163 continue; 2164 } 2165 size_t numsessionrefs = mAudioSessionRefs.size(); 2166 bool found = false; 2167 for (size_t k = 0; k < numsessionrefs; k++) { 2168 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2169 if (ref->mSessionid == sessionid) { 2170 ALOGV(" session %d still exists for %d with %d refs", 2171 sessionid, ref->mPid, ref->mCnt); 2172 found = true; 2173 break; 2174 } 2175 } 2176 if (!found) { 2177 Mutex::Autolock _l(t->mLock); 2178 // remove all effects from the chain 2179 while (ec->mEffects.size()) { 2180 sp<EffectModule> effect = ec->mEffects[0]; 2181 effect->unPin(); 2182 t->removeEffect_l(effect); 2183 if (effect->purgeHandles()) { 2184 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2185 } 2186 AudioSystem::unregisterEffect(effect->id()); 2187 } 2188 } 2189 } 2190 return; 2191} 2192 2193// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2194AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2195{ 2196 return mPlaybackThreads.valueFor(output).get(); 2197} 2198 2199// checkMixerThread_l() must be called with AudioFlinger::mLock held 2200AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2201{ 2202 PlaybackThread *thread = checkPlaybackThread_l(output); 2203 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2204} 2205 2206// checkRecordThread_l() must be called with AudioFlinger::mLock held 2207AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2208{ 2209 return mRecordThreads.valueFor(input).get(); 2210} 2211 2212uint32_t AudioFlinger::nextUniqueId() 2213{ 2214 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2215} 2216 2217AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2218{ 2219 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2220 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2221 AudioStreamOut *output = thread->getOutput(); 2222 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2223 return thread; 2224 } 2225 } 2226 return NULL; 2227} 2228 2229audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2230{ 2231 PlaybackThread *thread = primaryPlaybackThread_l(); 2232 2233 if (thread == NULL) { 2234 return 0; 2235 } 2236 2237 return thread->outDevice(); 2238} 2239 2240sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2241 int triggerSession, 2242 int listenerSession, 2243 sync_event_callback_t callBack, 2244 wp<RefBase> cookie) 2245{ 2246 Mutex::Autolock _l(mLock); 2247 2248 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2249 status_t playStatus = NAME_NOT_FOUND; 2250 status_t recStatus = NAME_NOT_FOUND; 2251 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2252 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2253 if (playStatus == NO_ERROR) { 2254 return event; 2255 } 2256 } 2257 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2258 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2259 if (recStatus == NO_ERROR) { 2260 return event; 2261 } 2262 } 2263 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2264 mPendingSyncEvents.add(event); 2265 } else { 2266 ALOGV("createSyncEvent() invalid event %d", event->type()); 2267 event.clear(); 2268 } 2269 return event; 2270} 2271 2272// ---------------------------------------------------------------------------- 2273// Effect management 2274// ---------------------------------------------------------------------------- 2275 2276 2277status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2278{ 2279 Mutex::Autolock _l(mLock); 2280 return EffectQueryNumberEffects(numEffects); 2281} 2282 2283status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2284{ 2285 Mutex::Autolock _l(mLock); 2286 return EffectQueryEffect(index, descriptor); 2287} 2288 2289status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2290 effect_descriptor_t *descriptor) const 2291{ 2292 Mutex::Autolock _l(mLock); 2293 return EffectGetDescriptor(pUuid, descriptor); 2294} 2295 2296 2297sp<IEffect> AudioFlinger::createEffect( 2298 effect_descriptor_t *pDesc, 2299 const sp<IEffectClient>& effectClient, 2300 int32_t priority, 2301 audio_io_handle_t io, 2302 int sessionId, 2303 status_t *status, 2304 int *id, 2305 int *enabled) 2306{ 2307 status_t lStatus = NO_ERROR; 2308 sp<EffectHandle> handle; 2309 effect_descriptor_t desc; 2310 2311 pid_t pid = IPCThreadState::self()->getCallingPid(); 2312 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2313 pid, effectClient.get(), priority, sessionId, io); 2314 2315 if (pDesc == NULL) { 2316 lStatus = BAD_VALUE; 2317 goto Exit; 2318 } 2319 2320 // check audio settings permission for global effects 2321 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2322 lStatus = PERMISSION_DENIED; 2323 goto Exit; 2324 } 2325 2326 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2327 // that can only be created by audio policy manager (running in same process) 2328 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2329 lStatus = PERMISSION_DENIED; 2330 goto Exit; 2331 } 2332 2333 { 2334 if (!EffectIsNullUuid(&pDesc->uuid)) { 2335 // if uuid is specified, request effect descriptor 2336 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2337 if (lStatus < 0) { 2338 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2339 goto Exit; 2340 } 2341 } else { 2342 // if uuid is not specified, look for an available implementation 2343 // of the required type in effect factory 2344 if (EffectIsNullUuid(&pDesc->type)) { 2345 ALOGW("createEffect() no effect type"); 2346 lStatus = BAD_VALUE; 2347 goto Exit; 2348 } 2349 uint32_t numEffects = 0; 2350 effect_descriptor_t d; 2351 d.flags = 0; // prevent compiler warning 2352 bool found = false; 2353 2354 lStatus = EffectQueryNumberEffects(&numEffects); 2355 if (lStatus < 0) { 2356 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2357 goto Exit; 2358 } 2359 for (uint32_t i = 0; i < numEffects; i++) { 2360 lStatus = EffectQueryEffect(i, &desc); 2361 if (lStatus < 0) { 2362 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2363 continue; 2364 } 2365 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2366 // If matching type found save effect descriptor. If the session is 2367 // 0 and the effect is not auxiliary, continue enumeration in case 2368 // an auxiliary version of this effect type is available 2369 found = true; 2370 d = desc; 2371 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2372 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2373 break; 2374 } 2375 } 2376 } 2377 if (!found) { 2378 lStatus = BAD_VALUE; 2379 ALOGW("createEffect() effect not found"); 2380 goto Exit; 2381 } 2382 // For same effect type, chose auxiliary version over insert version if 2383 // connect to output mix (Compliance to OpenSL ES) 2384 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2385 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2386 desc = d; 2387 } 2388 } 2389 2390 // Do not allow auxiliary effects on a session different from 0 (output mix) 2391 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2392 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2393 lStatus = INVALID_OPERATION; 2394 goto Exit; 2395 } 2396 2397 // check recording permission for visualizer 2398 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2399 !recordingAllowed()) { 2400 lStatus = PERMISSION_DENIED; 2401 goto Exit; 2402 } 2403 2404 // return effect descriptor 2405 *pDesc = desc; 2406 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2407 // if the output returned by getOutputForEffect() is removed before we lock the 2408 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2409 // and we will exit safely 2410 io = AudioSystem::getOutputForEffect(&desc); 2411 ALOGV("createEffect got output %d", io); 2412 } 2413 2414 Mutex::Autolock _l(mLock); 2415 2416 // If output is not specified try to find a matching audio session ID in one of the 2417 // output threads. 2418 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2419 // because of code checking output when entering the function. 2420 // Note: io is never 0 when creating an effect on an input 2421 if (io == AUDIO_IO_HANDLE_NONE) { 2422 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2423 // output must be specified by AudioPolicyManager when using session 2424 // AUDIO_SESSION_OUTPUT_STAGE 2425 lStatus = BAD_VALUE; 2426 goto Exit; 2427 } 2428 // look for the thread where the specified audio session is present 2429 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2430 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2431 io = mPlaybackThreads.keyAt(i); 2432 break; 2433 } 2434 } 2435 if (io == 0) { 2436 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2437 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2438 io = mRecordThreads.keyAt(i); 2439 break; 2440 } 2441 } 2442 } 2443 // If no output thread contains the requested session ID, default to 2444 // first output. The effect chain will be moved to the correct output 2445 // thread when a track with the same session ID is created 2446 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2447 io = mPlaybackThreads.keyAt(0); 2448 } 2449 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2450 } 2451 ThreadBase *thread = checkRecordThread_l(io); 2452 if (thread == NULL) { 2453 thread = checkPlaybackThread_l(io); 2454 if (thread == NULL) { 2455 ALOGE("createEffect() unknown output thread"); 2456 lStatus = BAD_VALUE; 2457 goto Exit; 2458 } 2459 } 2460 2461 sp<Client> client = registerPid(pid); 2462 2463 // create effect on selected output thread 2464 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2465 &desc, enabled, &lStatus); 2466 if (handle != 0 && id != NULL) { 2467 *id = handle->id(); 2468 } 2469 if (handle == 0) { 2470 // remove local strong reference to Client with mClientLock held 2471 Mutex::Autolock _cl(mClientLock); 2472 client.clear(); 2473 } 2474 } 2475 2476Exit: 2477 *status = lStatus; 2478 return handle; 2479} 2480 2481status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2482 audio_io_handle_t dstOutput) 2483{ 2484 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2485 sessionId, srcOutput, dstOutput); 2486 Mutex::Autolock _l(mLock); 2487 if (srcOutput == dstOutput) { 2488 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2489 return NO_ERROR; 2490 } 2491 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2492 if (srcThread == NULL) { 2493 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2494 return BAD_VALUE; 2495 } 2496 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2497 if (dstThread == NULL) { 2498 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2499 return BAD_VALUE; 2500 } 2501 2502 Mutex::Autolock _dl(dstThread->mLock); 2503 Mutex::Autolock _sl(srcThread->mLock); 2504 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2505} 2506 2507// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2508status_t AudioFlinger::moveEffectChain_l(int sessionId, 2509 AudioFlinger::PlaybackThread *srcThread, 2510 AudioFlinger::PlaybackThread *dstThread, 2511 bool reRegister) 2512{ 2513 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2514 sessionId, srcThread, dstThread); 2515 2516 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2517 if (chain == 0) { 2518 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2519 sessionId, srcThread); 2520 return INVALID_OPERATION; 2521 } 2522 2523 // Check whether the destination thread has a channel count of FCC_2, which is 2524 // currently required for (most) effects. Prevent moving the effect chain here rather 2525 // than disabling the addEffect_l() call in dstThread below. 2526 if (dstThread->mChannelCount != FCC_2) { 2527 ALOGW("moveEffectChain_l() effect chain failed because" 2528 " destination thread %p channel count(%u) != %u", 2529 dstThread, dstThread->mChannelCount, FCC_2); 2530 return INVALID_OPERATION; 2531 } 2532 2533 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2534 // so that a new chain is created with correct parameters when first effect is added. This is 2535 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2536 // removed. 2537 srcThread->removeEffectChain_l(chain); 2538 2539 // transfer all effects one by one so that new effect chain is created on new thread with 2540 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2541 sp<EffectChain> dstChain; 2542 uint32_t strategy = 0; // prevent compiler warning 2543 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2544 Vector< sp<EffectModule> > removed; 2545 status_t status = NO_ERROR; 2546 while (effect != 0) { 2547 srcThread->removeEffect_l(effect); 2548 removed.add(effect); 2549 status = dstThread->addEffect_l(effect); 2550 if (status != NO_ERROR) { 2551 break; 2552 } 2553 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2554 if (effect->state() == EffectModule::ACTIVE || 2555 effect->state() == EffectModule::STOPPING) { 2556 effect->start(); 2557 } 2558 // if the move request is not received from audio policy manager, the effect must be 2559 // re-registered with the new strategy and output 2560 if (dstChain == 0) { 2561 dstChain = effect->chain().promote(); 2562 if (dstChain == 0) { 2563 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2564 status = NO_INIT; 2565 break; 2566 } 2567 strategy = dstChain->strategy(); 2568 } 2569 if (reRegister) { 2570 AudioSystem::unregisterEffect(effect->id()); 2571 AudioSystem::registerEffect(&effect->desc(), 2572 dstThread->id(), 2573 strategy, 2574 sessionId, 2575 effect->id()); 2576 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2577 } 2578 effect = chain->getEffectFromId_l(0); 2579 } 2580 2581 if (status != NO_ERROR) { 2582 for (size_t i = 0; i < removed.size(); i++) { 2583 srcThread->addEffect_l(removed[i]); 2584 if (dstChain != 0 && reRegister) { 2585 AudioSystem::unregisterEffect(removed[i]->id()); 2586 AudioSystem::registerEffect(&removed[i]->desc(), 2587 srcThread->id(), 2588 strategy, 2589 sessionId, 2590 removed[i]->id()); 2591 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2592 } 2593 } 2594 } 2595 2596 return status; 2597} 2598 2599bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2600{ 2601 if (mGlobalEffectEnableTime != 0 && 2602 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2603 return true; 2604 } 2605 2606 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2607 sp<EffectChain> ec = 2608 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2609 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2610 return true; 2611 } 2612 } 2613 return false; 2614} 2615 2616void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2617{ 2618 Mutex::Autolock _l(mLock); 2619 2620 mGlobalEffectEnableTime = systemTime(); 2621 2622 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2623 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2624 if (t->mType == ThreadBase::OFFLOAD) { 2625 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2626 } 2627 } 2628 2629} 2630 2631struct Entry { 2632#define MAX_NAME 32 // %Y%m%d%H%M%S_%d.wav 2633 char mName[MAX_NAME]; 2634}; 2635 2636int comparEntry(const void *p1, const void *p2) 2637{ 2638 return strcmp(((const Entry *) p1)->mName, ((const Entry *) p2)->mName); 2639} 2640 2641#ifdef TEE_SINK 2642void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2643{ 2644 NBAIO_Source *teeSource = source.get(); 2645 if (teeSource != NULL) { 2646 // .wav rotation 2647 // There is a benign race condition if 2 threads call this simultaneously. 2648 // They would both traverse the directory, but the result would simply be 2649 // failures at unlink() which are ignored. It's also unlikely since 2650 // normally dumpsys is only done by bugreport or from the command line. 2651 char teePath[32+256]; 2652 strcpy(teePath, "/data/misc/media"); 2653 size_t teePathLen = strlen(teePath); 2654 DIR *dir = opendir(teePath); 2655 teePath[teePathLen++] = '/'; 2656 if (dir != NULL) { 2657#define MAX_SORT 20 // number of entries to sort 2658#define MAX_KEEP 10 // number of entries to keep 2659 struct Entry entries[MAX_SORT]; 2660 size_t entryCount = 0; 2661 while (entryCount < MAX_SORT) { 2662 struct dirent de; 2663 struct dirent *result = NULL; 2664 int rc = readdir_r(dir, &de, &result); 2665 if (rc != 0) { 2666 ALOGW("readdir_r failed %d", rc); 2667 break; 2668 } 2669 if (result == NULL) { 2670 break; 2671 } 2672 if (result != &de) { 2673 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2674 break; 2675 } 2676 // ignore non .wav file entries 2677 size_t nameLen = strlen(de.d_name); 2678 if (nameLen <= 4 || nameLen >= MAX_NAME || 2679 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2680 continue; 2681 } 2682 strcpy(entries[entryCount++].mName, de.d_name); 2683 } 2684 (void) closedir(dir); 2685 if (entryCount > MAX_KEEP) { 2686 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2687 for (size_t i = 0; i < entryCount - MAX_KEEP; ++i) { 2688 strcpy(&teePath[teePathLen], entries[i].mName); 2689 (void) unlink(teePath); 2690 } 2691 } 2692 } else { 2693 if (fd >= 0) { 2694 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2695 } 2696 } 2697 char teeTime[16]; 2698 struct timeval tv; 2699 gettimeofday(&tv, NULL); 2700 struct tm tm; 2701 localtime_r(&tv.tv_sec, &tm); 2702 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2703 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2704 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2705 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2706 if (teeFd >= 0) { 2707 // FIXME use libsndfile 2708 char wavHeader[44]; 2709 memcpy(wavHeader, 2710 "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", 2711 sizeof(wavHeader)); 2712 NBAIO_Format format = teeSource->format(); 2713 unsigned channelCount = Format_channelCount(format); 2714 uint32_t sampleRate = Format_sampleRate(format); 2715 size_t frameSize = Format_frameSize(format); 2716 wavHeader[22] = channelCount; // number of channels 2717 wavHeader[24] = sampleRate; // sample rate 2718 wavHeader[25] = sampleRate >> 8; 2719 wavHeader[32] = frameSize; // block alignment 2720 wavHeader[33] = frameSize >> 8; 2721 write(teeFd, wavHeader, sizeof(wavHeader)); 2722 size_t total = 0; 2723 bool firstRead = true; 2724#define TEE_SINK_READ 1024 // frames per I/O operation 2725 void *buffer = malloc(TEE_SINK_READ * frameSize); 2726 for (;;) { 2727 size_t count = TEE_SINK_READ; 2728 ssize_t actual = teeSource->read(buffer, count, 2729 AudioBufferProvider::kInvalidPTS); 2730 bool wasFirstRead = firstRead; 2731 firstRead = false; 2732 if (actual <= 0) { 2733 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2734 continue; 2735 } 2736 break; 2737 } 2738 ALOG_ASSERT(actual <= (ssize_t)count); 2739 write(teeFd, buffer, actual * frameSize); 2740 total += actual; 2741 } 2742 free(buffer); 2743 lseek(teeFd, (off_t) 4, SEEK_SET); 2744 uint32_t temp = 44 + total * frameSize - 8; 2745 // FIXME not big-endian safe 2746 write(teeFd, &temp, sizeof(temp)); 2747 lseek(teeFd, (off_t) 40, SEEK_SET); 2748 temp = total * frameSize; 2749 // FIXME not big-endian safe 2750 write(teeFd, &temp, sizeof(temp)); 2751 close(teeFd); 2752 if (fd >= 0) { 2753 dprintf(fd, "tee copied to %s\n", teePath); 2754 } 2755 } else { 2756 if (fd >= 0) { 2757 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2758 } 2759 } 2760 } 2761} 2762#endif 2763 2764// ---------------------------------------------------------------------------- 2765 2766status_t AudioFlinger::onTransact( 2767 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2768{ 2769 return BnAudioFlinger::onTransact(code, data, reply, flags); 2770} 2771 2772}; // namespace android 2773