AudioFlinger.h revision 022b9953153bdb1984f0abb17d21ef8c1826ad49
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#ifndef ANDROID_AUDIO_FLINGER_H 19#define ANDROID_AUDIO_FLINGER_H 20 21#include "Configuration.h" 22#include <deque> 23#include <map> 24#include <stdint.h> 25#include <sys/types.h> 26#include <limits.h> 27 28#include <cutils/compiler.h> 29#include <cutils/properties.h> 30 31#include <media/IAudioFlinger.h> 32#include <media/IAudioFlingerClient.h> 33#include <media/IAudioTrack.h> 34#include <media/IAudioRecord.h> 35#include <media/AudioSystem.h> 36#include <media/AudioTrack.h> 37 38#include <utils/Atomic.h> 39#include <utils/Errors.h> 40#include <utils/threads.h> 41#include <utils/SortedVector.h> 42#include <utils/TypeHelpers.h> 43#include <utils/Vector.h> 44 45#include <binder/BinderService.h> 46#include <binder/MemoryDealer.h> 47 48#include <system/audio.h> 49#include <system/audio_policy.h> 50 51#include <media/audiohal/EffectBufferHalInterface.h> 52#include <media/audiohal/StreamHalInterface.h> 53#include <media/AudioBufferProvider.h> 54#include <media/ExtendedAudioBufferProvider.h> 55 56#include "FastCapture.h" 57#include "FastMixer.h" 58#include <media/nbaio/NBAIO.h> 59#include "AudioWatchdog.h" 60#include "AudioMixer.h" 61#include "AudioStreamOut.h" 62#include "SpdifStreamOut.h" 63#include "AudioHwDevice.h" 64#include "LinearMap.h" 65 66#include <powermanager/IPowerManager.h> 67 68#include <media/nbaio/NBLog.h> 69#include <private/media/AudioTrackShared.h> 70 71namespace android { 72 73struct audio_track_cblk_t; 74struct effect_param_cblk_t; 75class AudioMixer; 76class AudioBuffer; 77class AudioResampler; 78class DeviceHalInterface; 79class DevicesFactoryHalInterface; 80class EffectsFactoryHalInterface; 81class FastMixer; 82class PassthruBufferProvider; 83class ServerProxy; 84 85// ---------------------------------------------------------------------------- 86 87static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 88 89 90// Max shared memory size for audio tracks and audio records per client process 91static const size_t kClientSharedHeapSizeBytes = 1024*1024; 92// Shared memory size multiplier for non low ram devices 93static const size_t kClientSharedHeapSizeMultiplier = 4; 94 95#define INCLUDING_FROM_AUDIOFLINGER_H 96 97class AudioFlinger : 98 public BinderService<AudioFlinger>, 99 public BnAudioFlinger 100{ 101 friend class BinderService<AudioFlinger>; // for AudioFlinger() 102public: 103 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 104 105 virtual status_t dump(int fd, const Vector<String16>& args); 106 107 // IAudioFlinger interface, in binder opcode order 108 virtual sp<IAudioTrack> createTrack( 109 audio_stream_type_t streamType, 110 uint32_t sampleRate, 111 audio_format_t format, 112 audio_channel_mask_t channelMask, 113 size_t *pFrameCount, 114 audio_output_flags_t *flags, 115 const sp<IMemory>& sharedBuffer, 116 audio_io_handle_t output, 117 pid_t pid, 118 pid_t tid, 119 audio_session_t *sessionId, 120 int clientUid, 121 status_t *status /*non-NULL*/, 122 audio_port_handle_t portId); 123 124 virtual sp<IAudioRecord> openRecord( 125 audio_io_handle_t input, 126 uint32_t sampleRate, 127 audio_format_t format, 128 audio_channel_mask_t channelMask, 129 const String16& opPackageName, 130 size_t *pFrameCount, 131 audio_input_flags_t *flags, 132 pid_t pid, 133 pid_t tid, 134 int clientUid, 135 audio_session_t *sessionId, 136 size_t *notificationFrames, 137 sp<IMemory>& cblk, 138 sp<IMemory>& buffers, 139 status_t *status /*non-NULL*/, 140 audio_port_handle_t portId); 141 142 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 143 virtual audio_format_t format(audio_io_handle_t output) const; 144 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 145 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 146 virtual uint32_t latency(audio_io_handle_t output) const; 147 148 virtual status_t setMasterVolume(float value); 149 virtual status_t setMasterMute(bool muted); 150 151 virtual float masterVolume() const; 152 virtual bool masterMute() const; 153 154 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 155 audio_io_handle_t output); 156 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 157 158 virtual float streamVolume(audio_stream_type_t stream, 159 audio_io_handle_t output) const; 160 virtual bool streamMute(audio_stream_type_t stream) const; 161 162 virtual status_t setMode(audio_mode_t mode); 163 164 virtual status_t setMicMute(bool state); 165 virtual bool getMicMute() const; 166 167 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 168 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 169 170 virtual void registerClient(const sp<IAudioFlingerClient>& client); 171 172 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 173 audio_channel_mask_t channelMask) const; 174 175 virtual status_t openOutput(audio_module_handle_t module, 176 audio_io_handle_t *output, 177 audio_config_t *config, 178 audio_devices_t *devices, 179 const String8& address, 180 uint32_t *latencyMs, 181 audio_output_flags_t flags); 182 183 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 184 audio_io_handle_t output2); 185 186 virtual status_t closeOutput(audio_io_handle_t output); 187 188 virtual status_t suspendOutput(audio_io_handle_t output); 189 190 virtual status_t restoreOutput(audio_io_handle_t output); 191 192 virtual status_t openInput(audio_module_handle_t module, 193 audio_io_handle_t *input, 194 audio_config_t *config, 195 audio_devices_t *device, 196 const String8& address, 197 audio_source_t source, 198 audio_input_flags_t flags); 199 200 virtual status_t closeInput(audio_io_handle_t input); 201 202 virtual status_t invalidateStream(audio_stream_type_t stream); 203 204 virtual status_t setVoiceVolume(float volume); 205 206 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 207 audio_io_handle_t output) const; 208 209 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 210 211 // This is the binder API. For the internal API see nextUniqueId(). 212 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 213 214 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 215 216 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 217 218 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 219 220 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 221 222 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 223 effect_descriptor_t *descriptor) const; 224 225 virtual sp<IEffect> createEffect( 226 effect_descriptor_t *pDesc, 227 const sp<IEffectClient>& effectClient, 228 int32_t priority, 229 audio_io_handle_t io, 230 audio_session_t sessionId, 231 const String16& opPackageName, 232 pid_t pid, 233 status_t *status /*non-NULL*/, 234 int *id, 235 int *enabled); 236 237 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 238 audio_io_handle_t dstOutput); 239 240 virtual audio_module_handle_t loadHwModule(const char *name); 241 242 virtual uint32_t getPrimaryOutputSamplingRate(); 243 virtual size_t getPrimaryOutputFrameCount(); 244 245 virtual status_t setLowRamDevice(bool isLowRamDevice); 246 247 /* List available audio ports and their attributes */ 248 virtual status_t listAudioPorts(unsigned int *num_ports, 249 struct audio_port *ports); 250 251 /* Get attributes for a given audio port */ 252 virtual status_t getAudioPort(struct audio_port *port); 253 254 /* Create an audio patch between several source and sink ports */ 255 virtual status_t createAudioPatch(const struct audio_patch *patch, 256 audio_patch_handle_t *handle); 257 258 /* Release an audio patch */ 259 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 260 261 /* List existing audio patches */ 262 virtual status_t listAudioPatches(unsigned int *num_patches, 263 struct audio_patch *patches); 264 265 /* Set audio port configuration */ 266 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 267 268 /* Get the HW synchronization source used for an audio session */ 269 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 270 271 /* Indicate JAVA services are ready (scheduling, power management ...) */ 272 virtual status_t systemReady(); 273 274 virtual status_t onTransact( 275 uint32_t code, 276 const Parcel& data, 277 Parcel* reply, 278 uint32_t flags); 279 280 // end of IAudioFlinger interface 281 282 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 283 void unregisterWriter(const sp<NBLog::Writer>& writer); 284 sp<EffectsFactoryHalInterface> getEffectsFactory(); 285private: 286 static const size_t kLogMemorySize = 40 * 1024; 287 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 288 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 289 // for as long as possible. The memory is only freed when it is needed for another log writer. 290 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 291 Mutex mUnregisteredWritersLock; 292public: 293 294 class SyncEvent; 295 296 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 297 298 class SyncEvent : public RefBase { 299 public: 300 SyncEvent(AudioSystem::sync_event_t type, 301 audio_session_t triggerSession, 302 audio_session_t listenerSession, 303 sync_event_callback_t callBack, 304 wp<RefBase> cookie) 305 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 306 mCallback(callBack), mCookie(cookie) 307 {} 308 309 virtual ~SyncEvent() {} 310 311 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 312 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 313 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 314 AudioSystem::sync_event_t type() const { return mType; } 315 audio_session_t triggerSession() const { return mTriggerSession; } 316 audio_session_t listenerSession() const { return mListenerSession; } 317 wp<RefBase> cookie() const { return mCookie; } 318 319 private: 320 const AudioSystem::sync_event_t mType; 321 const audio_session_t mTriggerSession; 322 const audio_session_t mListenerSession; 323 sync_event_callback_t mCallback; 324 const wp<RefBase> mCookie; 325 mutable Mutex mLock; 326 }; 327 328 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 329 audio_session_t triggerSession, 330 audio_session_t listenerSession, 331 sync_event_callback_t callBack, 332 const wp<RefBase>& cookie); 333 334private: 335 336 audio_mode_t getMode() const { return mMode; } 337 338 bool btNrecIsOff() const { return mBtNrecIsOff; } 339 340 AudioFlinger() ANDROID_API; 341 virtual ~AudioFlinger(); 342 343 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 344 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 345 NO_INIT : NO_ERROR; } 346 347 // RefBase 348 virtual void onFirstRef(); 349 350 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 351 audio_devices_t devices); 352 void purgeStaleEffects_l(); 353 354 // Set kEnableExtendedChannels to true to enable greater than stereo output 355 // for the MixerThread and device sink. Number of channels allowed is 356 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 357 static const bool kEnableExtendedChannels = true; 358 359 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 360 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 361 switch (audio_channel_mask_get_representation(channelMask)) { 362 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 363 uint32_t channelCount = FCC_2; // stereo is default 364 if (kEnableExtendedChannels) { 365 channelCount = audio_channel_count_from_out_mask(channelMask); 366 if (channelCount < FCC_2 // mono is not supported at this time 367 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 368 return false; 369 } 370 } 371 // check that channelMask is the "canonical" one we expect for the channelCount. 372 return channelMask == audio_channel_out_mask_from_count(channelCount); 373 } 374 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 375 if (kEnableExtendedChannels) { 376 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 377 if (channelCount >= FCC_2 // mono is not supported at this time 378 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 379 return true; 380 } 381 } 382 return false; 383 default: 384 return false; 385 } 386 } 387 388 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 389 static const bool kEnableExtendedPrecision = true; 390 391 // Returns true if format is permitted for the PCM sink in the MixerThread 392 static inline bool isValidPcmSinkFormat(audio_format_t format) { 393 switch (format) { 394 case AUDIO_FORMAT_PCM_16_BIT: 395 return true; 396 case AUDIO_FORMAT_PCM_FLOAT: 397 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 398 case AUDIO_FORMAT_PCM_32_BIT: 399 case AUDIO_FORMAT_PCM_8_24_BIT: 400 return kEnableExtendedPrecision; 401 default: 402 return false; 403 } 404 } 405 406 // standby delay for MIXER and DUPLICATING playback threads is read from property 407 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 408 static nsecs_t mStandbyTimeInNsecs; 409 410 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 411 // AudioFlinger::setParameters() updates, other threads read w/o lock 412 static uint32_t mScreenState; 413 414 // Internal dump utilities. 415 static const int kDumpLockRetries = 50; 416 static const int kDumpLockSleepUs = 20000; 417 static bool dumpTryLock(Mutex& mutex); 418 void dumpPermissionDenial(int fd, const Vector<String16>& args); 419 void dumpClients(int fd, const Vector<String16>& args); 420 void dumpInternals(int fd, const Vector<String16>& args); 421 422 // --- Client --- 423 class Client : public RefBase { 424 public: 425 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 426 virtual ~Client(); 427 sp<MemoryDealer> heap() const; 428 pid_t pid() const { return mPid; } 429 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 430 431 private: 432 Client(const Client&); 433 Client& operator = (const Client&); 434 const sp<AudioFlinger> mAudioFlinger; 435 sp<MemoryDealer> mMemoryDealer; 436 const pid_t mPid; 437 }; 438 439 // --- Notification Client --- 440 class NotificationClient : public IBinder::DeathRecipient { 441 public: 442 NotificationClient(const sp<AudioFlinger>& audioFlinger, 443 const sp<IAudioFlingerClient>& client, 444 pid_t pid); 445 virtual ~NotificationClient(); 446 447 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 448 449 // IBinder::DeathRecipient 450 virtual void binderDied(const wp<IBinder>& who); 451 452 private: 453 NotificationClient(const NotificationClient&); 454 NotificationClient& operator = (const NotificationClient&); 455 456 const sp<AudioFlinger> mAudioFlinger; 457 const pid_t mPid; 458 const sp<IAudioFlingerClient> mAudioFlingerClient; 459 }; 460 461 class TrackHandle; 462 class RecordHandle; 463 class RecordThread; 464 class PlaybackThread; 465 class MixerThread; 466 class DirectOutputThread; 467 class OffloadThread; 468 class DuplicatingThread; 469 class AsyncCallbackThread; 470 class Track; 471 class RecordTrack; 472 class EffectModule; 473 class EffectHandle; 474 class EffectChain; 475 476 struct AudioStreamIn; 477 478 struct stream_type_t { 479 stream_type_t() 480 : volume(1.0f), 481 mute(false) 482 { 483 } 484 float volume; 485 bool mute; 486 }; 487 488 // --- PlaybackThread --- 489 490#include "Threads.h" 491 492#include "Effects.h" 493 494#include "PatchPanel.h" 495 496 // server side of the client's IAudioTrack 497 class TrackHandle : public android::BnAudioTrack { 498 public: 499 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 500 virtual ~TrackHandle(); 501 virtual sp<IMemory> getCblk() const; 502 virtual status_t start(); 503 virtual void stop(); 504 virtual void flush(); 505 virtual void pause(); 506 virtual status_t attachAuxEffect(int effectId); 507 virtual status_t setParameters(const String8& keyValuePairs); 508 virtual status_t getTimestamp(AudioTimestamp& timestamp); 509 virtual void signal(); // signal playback thread for a change in control block 510 511 virtual status_t onTransact( 512 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 513 514 private: 515 const sp<PlaybackThread::Track> mTrack; 516 }; 517 518 // server side of the client's IAudioRecord 519 class RecordHandle : public android::BnAudioRecord { 520 public: 521 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 522 virtual ~RecordHandle(); 523 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 524 audio_session_t triggerSession); 525 virtual void stop(); 526 virtual status_t onTransact( 527 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 528 private: 529 const sp<RecordThread::RecordTrack> mRecordTrack; 530 531 // for use from destructor 532 void stop_nonvirtual(); 533 }; 534 535 536 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 537 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 538 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 539 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 540 sp<RecordThread> openInput_l(audio_module_handle_t module, 541 audio_io_handle_t *input, 542 audio_config_t *config, 543 audio_devices_t device, 544 const String8& address, 545 audio_source_t source, 546 audio_input_flags_t flags); 547 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 548 audio_io_handle_t *output, 549 audio_config_t *config, 550 audio_devices_t devices, 551 const String8& address, 552 audio_output_flags_t flags); 553 554 void closeOutputFinish(const sp<PlaybackThread>& thread); 555 void closeInputFinish(const sp<RecordThread>& thread); 556 557 // no range check, AudioFlinger::mLock held 558 bool streamMute_l(audio_stream_type_t stream) const 559 { return mStreamTypes[stream].mute; } 560 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 561 float streamVolume_l(audio_stream_type_t stream) const 562 { return mStreamTypes[stream].volume; } 563 void ioConfigChanged(audio_io_config_event event, 564 const sp<AudioIoDescriptor>& ioDesc, 565 pid_t pid = 0); 566 567 // Allocate an audio_unique_id_t. 568 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 569 // audio_module_handle_t, and audio_patch_handle_t. 570 // They all share the same ID space, but the namespaces are actually independent 571 // because there are separate KeyedVectors for each kind of ID. 572 // The return value is cast to the specific type depending on how the ID will be used. 573 // FIXME This API does not handle rollover to zero (for unsigned IDs), 574 // or from positive to negative (for signed IDs). 575 // Thus it may fail by returning an ID of the wrong sign, 576 // or by returning a non-unique ID. 577 // This is the internal API. For the binder API see newAudioUniqueId(). 578 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 579 580 status_t moveEffectChain_l(audio_session_t sessionId, 581 PlaybackThread *srcThread, 582 PlaybackThread *dstThread, 583 bool reRegister); 584 585 // return thread associated with primary hardware device, or NULL 586 PlaybackThread *primaryPlaybackThread_l() const; 587 audio_devices_t primaryOutputDevice_l() const; 588 589 // return the playback thread with smallest HAL buffer size, and prefer fast 590 PlaybackThread *fastPlaybackThread_l() const; 591 592 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 593 594 595 void removeClient_l(pid_t pid); 596 void removeNotificationClient(pid_t pid); 597 bool isNonOffloadableGlobalEffectEnabled_l(); 598 void onNonOffloadableGlobalEffectEnable(); 599 bool isSessionAcquired_l(audio_session_t audioSession); 600 601 // Store an effect chain to mOrphanEffectChains keyed vector. 602 // Called when a thread exits and effects are still attached to it. 603 // If effects are later created on the same session, they will reuse the same 604 // effect chain and same instances in the effect library. 605 // return ALREADY_EXISTS if a chain with the same session already exists in 606 // mOrphanEffectChains. Note that this should never happen as there is only one 607 // chain for a given session and it is attached to only one thread at a time. 608 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 609 // Get an effect chain for the specified session in mOrphanEffectChains and remove 610 // it if found. Returns 0 if not found (this is the most common case). 611 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 612 // Called when the last effect handle on an effect instance is removed. If this 613 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 614 // and removed from mOrphanEffectChains if it does not contain any effect. 615 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 616 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 617 618 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 619 620 // AudioStreamIn is immutable, so their fields are const. 621 // For emphasis, we could also make all pointers to them be "const *", 622 // but that would clutter the code unnecessarily. 623 624 struct AudioStreamIn { 625 AudioHwDevice* const audioHwDev; 626 sp<StreamInHalInterface> stream; 627 audio_input_flags_t flags; 628 629 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 630 631 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 632 audioHwDev(dev), stream(in), flags(flags) {} 633 }; 634 635 // for mAudioSessionRefs only 636 struct AudioSessionRef { 637 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 638 mSessionid(sessionid), mPid(pid), mCnt(1) {} 639 const audio_session_t mSessionid; 640 const pid_t mPid; 641 int mCnt; 642 }; 643 644 mutable Mutex mLock; 645 // protects mClients and mNotificationClients. 646 // must be locked after mLock and ThreadBase::mLock if both must be locked 647 // avoids acquiring AudioFlinger::mLock from inside thread loop. 648 mutable Mutex mClientLock; 649 // protected by mClientLock 650 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 651 652 mutable Mutex mHardwareLock; 653 // NOTE: If both mLock and mHardwareLock mutexes must be held, 654 // always take mLock before mHardwareLock 655 656 // These two fields are immutable after onFirstRef(), so no lock needed to access 657 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 658 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 659 660 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 661 662 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 663 enum hardware_call_state { 664 AUDIO_HW_IDLE = 0, // no operation in progress 665 AUDIO_HW_INIT, // init_check 666 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 667 AUDIO_HW_OUTPUT_CLOSE, // unused 668 AUDIO_HW_INPUT_OPEN, // unused 669 AUDIO_HW_INPUT_CLOSE, // unused 670 AUDIO_HW_STANDBY, // unused 671 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 672 AUDIO_HW_GET_ROUTING, // unused 673 AUDIO_HW_SET_ROUTING, // unused 674 AUDIO_HW_GET_MODE, // unused 675 AUDIO_HW_SET_MODE, // set_mode 676 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 677 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 678 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 679 AUDIO_HW_SET_PARAMETER, // set_parameters 680 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 681 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 682 AUDIO_HW_GET_PARAMETER, // get_parameters 683 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 684 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 685 }; 686 687 mutable hardware_call_state mHardwareStatus; // for dump only 688 689 690 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 691 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 692 693 // member variables below are protected by mLock 694 float mMasterVolume; 695 bool mMasterMute; 696 // end of variables protected by mLock 697 698 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 699 700 // protected by mClientLock 701 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 702 703 // updated by atomic_fetch_add_explicit 704 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 705 706 audio_mode_t mMode; 707 bool mBtNrecIsOff; 708 709 // protected by mLock 710 Vector<AudioSessionRef*> mAudioSessionRefs; 711 712 float masterVolume_l() const; 713 bool masterMute_l() const; 714 audio_module_handle_t loadHwModule_l(const char *name); 715 716 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 717 // to be created 718 719 // Effect chains without a valid thread 720 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 721 722 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 723 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 724private: 725 sp<Client> registerPid(pid_t pid); // always returns non-0 726 727 // for use from destructor 728 status_t closeOutput_nonvirtual(audio_io_handle_t output); 729 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 730 status_t closeInput_nonvirtual(audio_io_handle_t input); 731 void closeInputInternal_l(const sp<RecordThread>& thread); 732 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 733 734 status_t checkStreamType(audio_stream_type_t stream) const; 735 736#ifdef TEE_SINK 737 // all record threads serially share a common tee sink, which is re-created on format change 738 sp<NBAIO_Sink> mRecordTeeSink; 739 sp<NBAIO_Source> mRecordTeeSource; 740#endif 741 742public: 743 744#ifdef TEE_SINK 745 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 746 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 747 748 // whether tee sink is enabled by property 749 static bool mTeeSinkInputEnabled; 750 static bool mTeeSinkOutputEnabled; 751 static bool mTeeSinkTrackEnabled; 752 753 // runtime configured size of each tee sink pipe, in frames 754 static size_t mTeeSinkInputFrames; 755 static size_t mTeeSinkOutputFrames; 756 static size_t mTeeSinkTrackFrames; 757 758 // compile-time default size of tee sink pipes, in frames 759 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 760 static const size_t kTeeSinkInputFramesDefault = 0x200000; 761 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 762 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 763#endif 764 765 // This method reads from a variable without mLock, but the variable is updated under mLock. So 766 // we might read a stale value, or a value that's inconsistent with respect to other variables. 767 // In this case, it's safe because the return value isn't used for making an important decision. 768 // The reason we don't want to take mLock is because it could block the caller for a long time. 769 bool isLowRamDevice() const { return mIsLowRamDevice; } 770 771private: 772 bool mIsLowRamDevice; 773 bool mIsDeviceTypeKnown; 774 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 775 776 sp<PatchPanel> mPatchPanel; 777 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 778 779 bool mSystemReady; 780}; 781 782#undef INCLUDING_FROM_AUDIOFLINGER_H 783 784std::string formatToString(audio_format_t format); 785std::string inputFlagsToString(audio_input_flags_t flags); 786std::string outputFlagsToString(audio_output_flags_t flags); 787std::string devicesToString(audio_devices_t devices); 788const char *sourceToString(audio_source_t source); 789 790// ---------------------------------------------------------------------------- 791 792} // namespace android 793 794#endif // ANDROID_AUDIO_FLINGER_H 795