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 <stdint.h> 23 #include <sys/types.h> 24 #include <limits.h> 25 26 #include <common_time/cc_helper.h> 27 28 #include <cutils/compiler.h> 29 30 #include <media/IAudioFlinger.h> 31 #include <media/IAudioFlingerClient.h> 32 #include <media/IAudioTrack.h> 33 #include <media/IAudioRecord.h> 34 #include <media/AudioSystem.h> 35 #include <media/AudioTrack.h> 36 37 #include <utils/Atomic.h> 38 #include <utils/Errors.h> 39 #include <utils/threads.h> 40 #include <utils/SortedVector.h> 41 #include <utils/TypeHelpers.h> 42 #include <utils/Vector.h> 43 44 #include <binder/BinderService.h> 45 #include <binder/MemoryDealer.h> 46 47 #include <system/audio.h> 48 #include <hardware/audio.h> 49 #include <hardware/audio_policy.h> 50 51 #include <media/AudioBufferProvider.h> 52 #include <media/ExtendedAudioBufferProvider.h> 53 54 #include "FastCapture.h" 55 #include "FastMixer.h" 56 #include <media/nbaio/NBAIO.h> 57 #include "AudioWatchdog.h" 58 #include "AudioMixer.h" 59 #include "AudioStreamOut.h" 60 #include "SpdifStreamOut.h" 61 #include "AudioHwDevice.h" 62 63 #include <powermanager/IPowerManager.h> 64 65 #include <media/nbaio/NBLog.h> 66 #include <private/media/AudioTrackShared.h> 67 68 namespace android { 69 70 struct audio_track_cblk_t; 71 struct effect_param_cblk_t; 72 class AudioMixer; 73 class AudioBuffer; 74 class AudioResampler; 75 class FastMixer; 76 class PassthruBufferProvider; 77 class ServerProxy; 78 79 // ---------------------------------------------------------------------------- 80 81 // The macro FCC_2 highlights some (but not all) places where there are are 2-channel assumptions. 82 // This is typically due to legacy implementation of stereo input or output. 83 // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 84 #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 85 // The macro FCC_8 highlights places where there are 8-channel assumptions. 86 // This is typically due to audio mixer and resampler limitations. 87 #define FCC_8 8 // FCC_8 = Fixed Channel Count 8 88 89 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 90 91 92 // Max shared memory size for audio tracks and audio records per client process 93 static const size_t kClientSharedHeapSizeBytes = 1024*1024; 94 // Shared memory size multiplier for non low ram devices 95 static const size_t kClientSharedHeapSizeMultiplier = 4; 96 97 #define INCLUDING_FROM_AUDIOFLINGER_H 98 99 class AudioFlinger : 100 public BinderService<AudioFlinger>, 101 public BnAudioFlinger 102 { 103 friend class BinderService<AudioFlinger>; // for AudioFlinger() 104 public: getServiceName()105 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 106 107 virtual status_t dump(int fd, const Vector<String16>& args); 108 109 // IAudioFlinger interface, in binder opcode order 110 virtual sp<IAudioTrack> createTrack( 111 audio_stream_type_t streamType, 112 uint32_t sampleRate, 113 audio_format_t format, 114 audio_channel_mask_t channelMask, 115 size_t *pFrameCount, 116 IAudioFlinger::track_flags_t *flags, 117 const sp<IMemory>& sharedBuffer, 118 audio_io_handle_t output, 119 pid_t tid, 120 int *sessionId, 121 int clientUid, 122 status_t *status /*non-NULL*/); 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 IAudioFlinger::track_flags_t *flags, 132 pid_t tid, 133 int clientUid, 134 int *sessionId, 135 size_t *notificationFrames, 136 sp<IMemory>& cblk, 137 sp<IMemory>& buffers, 138 status_t *status /*non-NULL*/); 139 140 virtual uint32_t sampleRate(audio_io_handle_t output) const; 141 virtual audio_format_t format(audio_io_handle_t output) const; 142 virtual size_t frameCount(audio_io_handle_t output) const; 143 virtual uint32_t latency(audio_io_handle_t output) const; 144 145 virtual status_t setMasterVolume(float value); 146 virtual status_t setMasterMute(bool muted); 147 148 virtual float masterVolume() const; 149 virtual bool masterMute() const; 150 151 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 152 audio_io_handle_t output); 153 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 154 155 virtual float streamVolume(audio_stream_type_t stream, 156 audio_io_handle_t output) const; 157 virtual bool streamMute(audio_stream_type_t stream) const; 158 159 virtual status_t setMode(audio_mode_t mode); 160 161 virtual status_t setMicMute(bool state); 162 virtual bool getMicMute() const; 163 164 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 165 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 166 167 virtual void registerClient(const sp<IAudioFlingerClient>& client); 168 169 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 170 audio_channel_mask_t channelMask) const; 171 172 virtual status_t openOutput(audio_module_handle_t module, 173 audio_io_handle_t *output, 174 audio_config_t *config, 175 audio_devices_t *devices, 176 const String8& address, 177 uint32_t *latencyMs, 178 audio_output_flags_t flags); 179 180 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 181 audio_io_handle_t output2); 182 183 virtual status_t closeOutput(audio_io_handle_t output); 184 185 virtual status_t suspendOutput(audio_io_handle_t output); 186 187 virtual status_t restoreOutput(audio_io_handle_t output); 188 189 virtual status_t openInput(audio_module_handle_t module, 190 audio_io_handle_t *input, 191 audio_config_t *config, 192 audio_devices_t *device, 193 const String8& address, 194 audio_source_t source, 195 audio_input_flags_t flags); 196 197 virtual status_t closeInput(audio_io_handle_t input); 198 199 virtual status_t invalidateStream(audio_stream_type_t stream); 200 201 virtual status_t setVoiceVolume(float volume); 202 203 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 204 audio_io_handle_t output) const; 205 206 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 207 208 virtual audio_unique_id_t newAudioUniqueId(); 209 210 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 211 212 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 213 214 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 215 216 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 217 218 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 219 effect_descriptor_t *descriptor) const; 220 221 virtual sp<IEffect> createEffect( 222 effect_descriptor_t *pDesc, 223 const sp<IEffectClient>& effectClient, 224 int32_t priority, 225 audio_io_handle_t io, 226 int sessionId, 227 const String16& opPackageName, 228 status_t *status /*non-NULL*/, 229 int *id, 230 int *enabled); 231 232 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 233 audio_io_handle_t dstOutput); 234 235 virtual audio_module_handle_t loadHwModule(const char *name); 236 237 virtual uint32_t getPrimaryOutputSamplingRate(); 238 virtual size_t getPrimaryOutputFrameCount(); 239 240 virtual status_t setLowRamDevice(bool isLowRamDevice); 241 242 /* List available audio ports and their attributes */ 243 virtual status_t listAudioPorts(unsigned int *num_ports, 244 struct audio_port *ports); 245 246 /* Get attributes for a given audio port */ 247 virtual status_t getAudioPort(struct audio_port *port); 248 249 /* Create an audio patch between several source and sink ports */ 250 virtual status_t createAudioPatch(const struct audio_patch *patch, 251 audio_patch_handle_t *handle); 252 253 /* Release an audio patch */ 254 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 255 256 /* List existing audio patches */ 257 virtual status_t listAudioPatches(unsigned int *num_patches, 258 struct audio_patch *patches); 259 260 /* Set audio port configuration */ 261 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 262 263 /* Get the HW synchronization source used for an audio session */ 264 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 265 266 /* Indicate JAVA services are ready (scheduling, power management ...) */ 267 virtual status_t systemReady(); 268 269 virtual status_t onTransact( 270 uint32_t code, 271 const Parcel& data, 272 Parcel* reply, 273 uint32_t flags); 274 275 // end of IAudioFlinger interface 276 277 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 278 void unregisterWriter(const sp<NBLog::Writer>& writer); 279 private: 280 static const size_t kLogMemorySize = 40 * 1024; 281 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 282 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 283 // for as long as possible. The memory is only freed when it is needed for another log writer. 284 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 285 Mutex mUnregisteredWritersLock; 286 public: 287 288 class SyncEvent; 289 290 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 291 292 class SyncEvent : public RefBase { 293 public: SyncEvent(AudioSystem::sync_event_t type,int triggerSession,int listenerSession,sync_event_callback_t callBack,wp<RefBase> cookie)294 SyncEvent(AudioSystem::sync_event_t type, 295 int triggerSession, 296 int listenerSession, 297 sync_event_callback_t callBack, 298 wp<RefBase> cookie) 299 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 300 mCallback(callBack), mCookie(cookie) 301 {} 302 ~SyncEvent()303 virtual ~SyncEvent() {} 304 trigger()305 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } isCancelled()306 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()307 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()308 AudioSystem::sync_event_t type() const { return mType; } triggerSession()309 int triggerSession() const { return mTriggerSession; } listenerSession()310 int listenerSession() const { return mListenerSession; } cookie()311 wp<RefBase> cookie() const { return mCookie; } 312 313 private: 314 const AudioSystem::sync_event_t mType; 315 const int mTriggerSession; 316 const int mListenerSession; 317 sync_event_callback_t mCallback; 318 const wp<RefBase> mCookie; 319 mutable Mutex mLock; 320 }; 321 322 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 323 int triggerSession, 324 int listenerSession, 325 sync_event_callback_t callBack, 326 wp<RefBase> cookie); 327 328 private: 329 getMode()330 audio_mode_t getMode() const { return mMode; } 331 btNrecIsOff()332 bool btNrecIsOff() const { return mBtNrecIsOff; } 333 334 AudioFlinger() ANDROID_API; 335 virtual ~AudioFlinger(); 336 337 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()338 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 339 NO_INIT : NO_ERROR; } 340 341 // RefBase 342 virtual void onFirstRef(); 343 344 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 345 audio_devices_t devices); 346 void purgeStaleEffects_l(); 347 348 // Set kEnableExtendedChannels to true to enable greater than stereo output 349 // for the MixerThread and device sink. Number of channels allowed is 350 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 351 static const bool kEnableExtendedChannels = true; 352 353 // Returns true if channel mask is permitted for the PCM sink in the MixerThread isValidPcmSinkChannelMask(audio_channel_mask_t channelMask)354 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 355 switch (audio_channel_mask_get_representation(channelMask)) { 356 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 357 uint32_t channelCount = FCC_2; // stereo is default 358 if (kEnableExtendedChannels) { 359 channelCount = audio_channel_count_from_out_mask(channelMask); 360 if (channelCount < FCC_2 // mono is not supported at this time 361 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 362 return false; 363 } 364 } 365 // check that channelMask is the "canonical" one we expect for the channelCount. 366 return channelMask == audio_channel_out_mask_from_count(channelCount); 367 } 368 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 369 if (kEnableExtendedChannels) { 370 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 371 if (channelCount >= FCC_2 // mono is not supported at this time 372 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 373 return true; 374 } 375 } 376 return false; 377 default: 378 return false; 379 } 380 } 381 382 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 383 static const bool kEnableExtendedPrecision = true; 384 385 // Returns true if format is permitted for the PCM sink in the MixerThread isValidPcmSinkFormat(audio_format_t format)386 static inline bool isValidPcmSinkFormat(audio_format_t format) { 387 switch (format) { 388 case AUDIO_FORMAT_PCM_16_BIT: 389 return true; 390 case AUDIO_FORMAT_PCM_FLOAT: 391 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 392 case AUDIO_FORMAT_PCM_32_BIT: 393 case AUDIO_FORMAT_PCM_8_24_BIT: 394 return kEnableExtendedPrecision; 395 default: 396 return false; 397 } 398 } 399 400 // standby delay for MIXER and DUPLICATING playback threads is read from property 401 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 402 static nsecs_t mStandbyTimeInNsecs; 403 404 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 405 // AudioFlinger::setParameters() updates, other threads read w/o lock 406 static uint32_t mScreenState; 407 408 // Internal dump utilities. 409 static const int kDumpLockRetries = 50; 410 static const int kDumpLockSleepUs = 20000; 411 static bool dumpTryLock(Mutex& mutex); 412 void dumpPermissionDenial(int fd, const Vector<String16>& args); 413 void dumpClients(int fd, const Vector<String16>& args); 414 void dumpInternals(int fd, const Vector<String16>& args); 415 416 // --- Client --- 417 class Client : public RefBase { 418 public: 419 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 420 virtual ~Client(); 421 sp<MemoryDealer> heap() const; pid()422 pid_t pid() const { return mPid; } audioFlinger()423 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 424 425 bool reserveTimedTrack(); 426 void releaseTimedTrack(); 427 428 private: 429 Client(const Client&); 430 Client& operator = (const Client&); 431 const sp<AudioFlinger> mAudioFlinger; 432 sp<MemoryDealer> mMemoryDealer; 433 const pid_t mPid; 434 435 Mutex mTimedTrackLock; 436 int mTimedTrackCount; 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 audioFlingerClient()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 { stream_type_tstream_type_t479 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 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 allocateTimedBuffer(size_t size, 508 sp<IMemory>* buffer); 509 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 510 int64_t pts); 511 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 512 int target); 513 virtual status_t setParameters(const String8& keyValuePairs); 514 virtual status_t getTimestamp(AudioTimestamp& timestamp); 515 virtual void signal(); // signal playback thread for a change in control block 516 517 virtual status_t onTransact( 518 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 519 520 private: 521 const sp<PlaybackThread::Track> mTrack; 522 }; 523 524 // server side of the client's IAudioRecord 525 class RecordHandle : public android::BnAudioRecord { 526 public: 527 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 528 virtual ~RecordHandle(); 529 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 530 virtual void stop(); 531 virtual status_t onTransact( 532 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 533 private: 534 const sp<RecordThread::RecordTrack> mRecordTrack; 535 536 // for use from destructor 537 void stop_nonvirtual(); 538 }; 539 540 541 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 542 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 543 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 544 sp<RecordThread> openInput_l(audio_module_handle_t module, 545 audio_io_handle_t *input, 546 audio_config_t *config, 547 audio_devices_t device, 548 const String8& address, 549 audio_source_t source, 550 audio_input_flags_t flags); 551 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 552 audio_io_handle_t *output, 553 audio_config_t *config, 554 audio_devices_t devices, 555 const String8& address, 556 audio_output_flags_t flags); 557 558 void closeOutputFinish(sp<PlaybackThread> thread); 559 void closeInputFinish(sp<RecordThread> thread); 560 561 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)562 bool streamMute_l(audio_stream_type_t stream) const 563 { return mStreamTypes[stream].mute; } 564 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held streamVolume_l(audio_stream_type_t stream)565 float streamVolume_l(audio_stream_type_t stream) const 566 { return mStreamTypes[stream].volume; } 567 void ioConfigChanged(audio_io_config_event event, 568 const sp<AudioIoDescriptor>& ioDesc, 569 pid_t pid = 0); 570 571 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 572 // They all share the same ID space, but the namespaces are actually independent 573 // because there are separate KeyedVectors for each kind of ID. 574 // The return value is uint32_t, but is cast to signed for some IDs. 575 // FIXME This API does not handle rollover to zero (for unsigned IDs), 576 // or from positive to negative (for signed IDs). 577 // Thus it may fail by returning an ID of the wrong sign, 578 // or by returning a non-unique ID. 579 uint32_t nextUniqueId(); 580 581 status_t moveEffectChain_l(int sessionId, 582 PlaybackThread *srcThread, 583 PlaybackThread *dstThread, 584 bool reRegister); 585 // return thread associated with primary hardware device, or NULL 586 PlaybackThread *primaryPlaybackThread_l() const; 587 audio_devices_t primaryOutputDevice_l() const; 588 589 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 590 591 592 void removeClient_l(pid_t pid); 593 void removeNotificationClient(pid_t pid); 594 bool isNonOffloadableGlobalEffectEnabled_l(); 595 void onNonOffloadableGlobalEffectEnable(); 596 597 // Store an effect chain to mOrphanEffectChains keyed vector. 598 // Called when a thread exits and effects are still attached to it. 599 // If effects are later created on the same session, they will reuse the same 600 // effect chain and same instances in the effect library. 601 // return ALREADY_EXISTS if a chain with the same session already exists in 602 // mOrphanEffectChains. Note that this should never happen as there is only one 603 // chain for a given session and it is attached to only one thread at a time. 604 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 605 // Get an effect chain for the specified session in mOrphanEffectChains and remove 606 // it if found. Returns 0 if not found (this is the most common case). 607 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 608 // Called when the last effect handle on an effect instance is removed. If this 609 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 610 // and removed from mOrphanEffectChains if it does not contain any effect. 611 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 612 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 613 614 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 615 616 // AudioStreamIn is immutable, so their fields are const. 617 // For emphasis, we could also make all pointers to them be "const *", 618 // but that would clutter the code unnecessarily. 619 620 struct AudioStreamIn { 621 AudioHwDevice* const audioHwDev; 622 audio_stream_in_t* const stream; 623 hwDevAudioStreamIn624 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 625 AudioStreamInAudioStreamIn626 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 627 audioHwDev(dev), stream(in) {} 628 }; 629 630 // for mAudioSessionRefs only 631 struct AudioSessionRef { AudioSessionRefAudioSessionRef632 AudioSessionRef(int sessionid, pid_t pid) : 633 mSessionid(sessionid), mPid(pid), mCnt(1) {} 634 const int mSessionid; 635 const pid_t mPid; 636 int mCnt; 637 }; 638 639 mutable Mutex mLock; 640 // protects mClients and mNotificationClients. 641 // must be locked after mLock and ThreadBase::mLock if both must be locked 642 // avoids acquiring AudioFlinger::mLock from inside thread loop. 643 mutable Mutex mClientLock; 644 // protected by mClientLock 645 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 646 647 mutable Mutex mHardwareLock; 648 // NOTE: If both mLock and mHardwareLock mutexes must be held, 649 // always take mLock before mHardwareLock 650 651 // These two fields are immutable after onFirstRef(), so no lock needed to access 652 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 653 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 654 655 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 656 enum hardware_call_state { 657 AUDIO_HW_IDLE = 0, // no operation in progress 658 AUDIO_HW_INIT, // init_check 659 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 660 AUDIO_HW_OUTPUT_CLOSE, // unused 661 AUDIO_HW_INPUT_OPEN, // unused 662 AUDIO_HW_INPUT_CLOSE, // unused 663 AUDIO_HW_STANDBY, // unused 664 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 665 AUDIO_HW_GET_ROUTING, // unused 666 AUDIO_HW_SET_ROUTING, // unused 667 AUDIO_HW_GET_MODE, // unused 668 AUDIO_HW_SET_MODE, // set_mode 669 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 670 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 671 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 672 AUDIO_HW_SET_PARAMETER, // set_parameters 673 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 674 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 675 AUDIO_HW_GET_PARAMETER, // get_parameters 676 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 677 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 678 }; 679 680 mutable hardware_call_state mHardwareStatus; // for dump only 681 682 683 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 684 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 685 686 // member variables below are protected by mLock 687 float mMasterVolume; 688 bool mMasterMute; 689 // end of variables protected by mLock 690 691 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 692 693 // protected by mClientLock 694 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 695 696 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 697 // nextUniqueId() returns uint32_t, but this is declared int32_t 698 // because the atomic operations require an int32_t 699 700 audio_mode_t mMode; 701 bool mBtNrecIsOff; 702 703 // protected by mLock 704 Vector<AudioSessionRef*> mAudioSessionRefs; 705 706 float masterVolume_l() const; 707 bool masterMute_l() const; 708 audio_module_handle_t loadHwModule_l(const char *name); 709 710 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 711 // to be created 712 713 // Effect chains without a valid thread 714 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 715 716 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 717 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 718 private: 719 sp<Client> registerPid(pid_t pid); // always returns non-0 720 721 // for use from destructor 722 status_t closeOutput_nonvirtual(audio_io_handle_t output); 723 void closeOutputInternal_l(sp<PlaybackThread> thread); 724 status_t closeInput_nonvirtual(audio_io_handle_t input); 725 void closeInputInternal_l(sp<RecordThread> thread); 726 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 727 728 status_t checkStreamType(audio_stream_type_t stream) const; 729 730 #ifdef TEE_SINK 731 // all record threads serially share a common tee sink, which is re-created on format change 732 sp<NBAIO_Sink> mRecordTeeSink; 733 sp<NBAIO_Source> mRecordTeeSource; 734 #endif 735 736 public: 737 738 #ifdef TEE_SINK 739 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 740 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 741 742 // whether tee sink is enabled by property 743 static bool mTeeSinkInputEnabled; 744 static bool mTeeSinkOutputEnabled; 745 static bool mTeeSinkTrackEnabled; 746 747 // runtime configured size of each tee sink pipe, in frames 748 static size_t mTeeSinkInputFrames; 749 static size_t mTeeSinkOutputFrames; 750 static size_t mTeeSinkTrackFrames; 751 752 // compile-time default size of tee sink pipes, in frames 753 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 754 static const size_t kTeeSinkInputFramesDefault = 0x200000; 755 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 756 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 757 #endif 758 759 // This method reads from a variable without mLock, but the variable is updated under mLock. So 760 // we might read a stale value, or a value that's inconsistent with respect to other variables. 761 // In this case, it's safe because the return value isn't used for making an important decision. 762 // The reason we don't want to take mLock is because it could block the caller for a long time. isLowRamDevice()763 bool isLowRamDevice() const { return mIsLowRamDevice; } 764 765 private: 766 bool mIsLowRamDevice; 767 bool mIsDeviceTypeKnown; 768 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 769 770 sp<PatchPanel> mPatchPanel; 771 772 bool mSystemReady; 773 }; 774 775 #undef INCLUDING_FROM_AUDIOFLINGER_H 776 777 const char *formatToString(audio_format_t format); 778 String8 inputFlagsToString(audio_input_flags_t flags); 779 String8 outputFlagsToString(audio_output_flags_t flags); 780 String8 devicesToString(audio_devices_t devices); 781 const char *sourceToString(audio_source_t source); 782 783 // ---------------------------------------------------------------------------- 784 785 } // namespace android 786 787 #endif // ANDROID_AUDIO_FLINGER_H 788