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 60 #include <powermanager/IPowerManager.h> 61 62 #include <media/nbaio/NBLog.h> 63 #include <private/media/AudioTrackShared.h> 64 65 namespace android { 66 67 struct audio_track_cblk_t; 68 struct effect_param_cblk_t; 69 class AudioMixer; 70 class AudioBuffer; 71 class AudioResampler; 72 class FastMixer; 73 class ServerProxy; 74 75 // ---------------------------------------------------------------------------- 76 77 // AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 78 // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 79 // Adding full support for > 2 channel capture or playback would require more than simply changing 80 // this #define. There is an independent hard-coded upper limit in AudioMixer; 81 // removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 82 // The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 83 // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 84 #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 85 86 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 87 88 #define INCLUDING_FROM_AUDIOFLINGER_H 89 90 class AudioFlinger : 91 public BinderService<AudioFlinger>, 92 public BnAudioFlinger 93 { 94 friend class BinderService<AudioFlinger>; // for AudioFlinger() 95 public: 96 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 97 98 virtual status_t dump(int fd, const Vector<String16>& args); 99 100 // IAudioFlinger interface, in binder opcode order 101 virtual sp<IAudioTrack> createTrack( 102 audio_stream_type_t streamType, 103 uint32_t sampleRate, 104 audio_format_t format, 105 audio_channel_mask_t channelMask, 106 size_t *pFrameCount, 107 IAudioFlinger::track_flags_t *flags, 108 const sp<IMemory>& sharedBuffer, 109 audio_io_handle_t output, 110 pid_t tid, 111 int *sessionId, 112 int clientUid, 113 status_t *status /*non-NULL*/); 114 115 virtual sp<IAudioRecord> openRecord( 116 audio_io_handle_t input, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 size_t *pFrameCount, 121 IAudioFlinger::track_flags_t *flags, 122 pid_t tid, 123 int *sessionId, 124 size_t *notificationFrames, 125 sp<IMemory>& cblk, 126 sp<IMemory>& buffers, 127 status_t *status /*non-NULL*/); 128 129 virtual uint32_t sampleRate(audio_io_handle_t output) const; 130 virtual audio_format_t format(audio_io_handle_t output) const; 131 virtual size_t frameCount(audio_io_handle_t output) const; 132 virtual uint32_t latency(audio_io_handle_t output) const; 133 134 virtual status_t setMasterVolume(float value); 135 virtual status_t setMasterMute(bool muted); 136 137 virtual float masterVolume() const; 138 virtual bool masterMute() const; 139 140 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 141 audio_io_handle_t output); 142 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 143 144 virtual float streamVolume(audio_stream_type_t stream, 145 audio_io_handle_t output) const; 146 virtual bool streamMute(audio_stream_type_t stream) const; 147 148 virtual status_t setMode(audio_mode_t mode); 149 150 virtual status_t setMicMute(bool state); 151 virtual bool getMicMute() const; 152 153 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 154 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 155 156 virtual void registerClient(const sp<IAudioFlingerClient>& client); 157 158 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 159 audio_channel_mask_t channelMask) const; 160 161 virtual status_t openOutput(audio_module_handle_t module, 162 audio_io_handle_t *output, 163 audio_config_t *config, 164 audio_devices_t *devices, 165 const String8& address, 166 uint32_t *latencyMs, 167 audio_output_flags_t flags); 168 169 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 170 audio_io_handle_t output2); 171 172 virtual status_t closeOutput(audio_io_handle_t output); 173 174 virtual status_t suspendOutput(audio_io_handle_t output); 175 176 virtual status_t restoreOutput(audio_io_handle_t output); 177 178 virtual status_t openInput(audio_module_handle_t module, 179 audio_io_handle_t *input, 180 audio_config_t *config, 181 audio_devices_t *device, 182 const String8& address, 183 audio_source_t source, 184 audio_input_flags_t flags); 185 186 virtual status_t closeInput(audio_io_handle_t input); 187 188 virtual status_t invalidateStream(audio_stream_type_t stream); 189 190 virtual status_t setVoiceVolume(float volume); 191 192 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 193 audio_io_handle_t output) const; 194 195 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 196 197 virtual audio_unique_id_t newAudioUniqueId(); 198 199 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 200 201 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 202 203 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 204 205 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 206 207 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 208 effect_descriptor_t *descriptor) const; 209 210 virtual sp<IEffect> createEffect( 211 effect_descriptor_t *pDesc, 212 const sp<IEffectClient>& effectClient, 213 int32_t priority, 214 audio_io_handle_t io, 215 int sessionId, 216 status_t *status /*non-NULL*/, 217 int *id, 218 int *enabled); 219 220 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 221 audio_io_handle_t dstOutput); 222 223 virtual audio_module_handle_t loadHwModule(const char *name); 224 225 virtual uint32_t getPrimaryOutputSamplingRate(); 226 virtual size_t getPrimaryOutputFrameCount(); 227 228 virtual status_t setLowRamDevice(bool isLowRamDevice); 229 230 /* List available audio ports and their attributes */ 231 virtual status_t listAudioPorts(unsigned int *num_ports, 232 struct audio_port *ports); 233 234 /* Get attributes for a given audio port */ 235 virtual status_t getAudioPort(struct audio_port *port); 236 237 /* Create an audio patch between several source and sink ports */ 238 virtual status_t createAudioPatch(const struct audio_patch *patch, 239 audio_patch_handle_t *handle); 240 241 /* Release an audio patch */ 242 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 243 244 /* List existing audio patches */ 245 virtual status_t listAudioPatches(unsigned int *num_patches, 246 struct audio_patch *patches); 247 248 /* Set audio port configuration */ 249 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 250 251 /* Get the HW synchronization source used for an audio session */ 252 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 253 254 virtual status_t onTransact( 255 uint32_t code, 256 const Parcel& data, 257 Parcel* reply, 258 uint32_t flags); 259 260 // end of IAudioFlinger interface 261 262 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 263 void unregisterWriter(const sp<NBLog::Writer>& writer); 264 private: 265 static const size_t kLogMemorySize = 40 * 1024; 266 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 267 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 268 // for as long as possible. The memory is only freed when it is needed for another log writer. 269 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 270 Mutex mUnregisteredWritersLock; 271 public: 272 273 class SyncEvent; 274 275 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 276 277 class SyncEvent : public RefBase { 278 public: 279 SyncEvent(AudioSystem::sync_event_t type, 280 int triggerSession, 281 int listenerSession, 282 sync_event_callback_t callBack, 283 wp<RefBase> cookie) 284 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 285 mCallback(callBack), mCookie(cookie) 286 {} 287 288 virtual ~SyncEvent() {} 289 290 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 291 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 292 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 293 AudioSystem::sync_event_t type() const { return mType; } 294 int triggerSession() const { return mTriggerSession; } 295 int listenerSession() const { return mListenerSession; } 296 wp<RefBase> cookie() const { return mCookie; } 297 298 private: 299 const AudioSystem::sync_event_t mType; 300 const int mTriggerSession; 301 const int mListenerSession; 302 sync_event_callback_t mCallback; 303 const wp<RefBase> mCookie; 304 mutable Mutex mLock; 305 }; 306 307 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 308 int triggerSession, 309 int listenerSession, 310 sync_event_callback_t callBack, 311 wp<RefBase> cookie); 312 313 private: 314 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 315 316 audio_mode_t getMode() const { return mMode; } 317 318 bool btNrecIsOff() const { return mBtNrecIsOff; } 319 320 AudioFlinger() ANDROID_API; 321 virtual ~AudioFlinger(); 322 323 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 324 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 325 NO_INIT : NO_ERROR; } 326 327 // RefBase 328 virtual void onFirstRef(); 329 330 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 331 audio_devices_t devices); 332 void purgeStaleEffects_l(); 333 334 // Set kEnableExtendedChannels to true to enable greater than stereo output 335 // for the MixerThread and device sink. Number of channels allowed is 336 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 337 static const bool kEnableExtendedChannels = true; 338 339 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 340 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 341 switch (audio_channel_mask_get_representation(channelMask)) { 342 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 343 uint32_t channelCount = FCC_2; // stereo is default 344 if (kEnableExtendedChannels) { 345 channelCount = audio_channel_count_from_out_mask(channelMask); 346 if (channelCount < FCC_2 // mono is not supported at this time 347 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 348 return false; 349 } 350 } 351 // check that channelMask is the "canonical" one we expect for the channelCount. 352 return channelMask == audio_channel_out_mask_from_count(channelCount); 353 } 354 default: 355 return false; 356 } 357 } 358 359 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 360 static const bool kEnableExtendedPrecision = true; 361 362 // Returns true if format is permitted for the PCM sink in the MixerThread 363 static inline bool isValidPcmSinkFormat(audio_format_t format) { 364 switch (format) { 365 case AUDIO_FORMAT_PCM_16_BIT: 366 return true; 367 case AUDIO_FORMAT_PCM_FLOAT: 368 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 369 case AUDIO_FORMAT_PCM_32_BIT: 370 case AUDIO_FORMAT_PCM_8_24_BIT: 371 return kEnableExtendedPrecision; 372 default: 373 return false; 374 } 375 } 376 377 // standby delay for MIXER and DUPLICATING playback threads is read from property 378 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 379 static nsecs_t mStandbyTimeInNsecs; 380 381 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 382 // AudioFlinger::setParameters() updates, other threads read w/o lock 383 static uint32_t mScreenState; 384 385 // Internal dump utilities. 386 static const int kDumpLockRetries = 50; 387 static const int kDumpLockSleepUs = 20000; 388 static bool dumpTryLock(Mutex& mutex); 389 void dumpPermissionDenial(int fd, const Vector<String16>& args); 390 void dumpClients(int fd, const Vector<String16>& args); 391 void dumpInternals(int fd, const Vector<String16>& args); 392 393 // --- Client --- 394 class Client : public RefBase { 395 public: 396 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 397 virtual ~Client(); 398 sp<MemoryDealer> heap() const; 399 pid_t pid() const { return mPid; } 400 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 401 402 bool reserveTimedTrack(); 403 void releaseTimedTrack(); 404 405 private: 406 Client(const Client&); 407 Client& operator = (const Client&); 408 const sp<AudioFlinger> mAudioFlinger; 409 const sp<MemoryDealer> mMemoryDealer; 410 const pid_t mPid; 411 412 Mutex mTimedTrackLock; 413 int mTimedTrackCount; 414 }; 415 416 // --- Notification Client --- 417 class NotificationClient : public IBinder::DeathRecipient { 418 public: 419 NotificationClient(const sp<AudioFlinger>& audioFlinger, 420 const sp<IAudioFlingerClient>& client, 421 pid_t pid); 422 virtual ~NotificationClient(); 423 424 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 425 426 // IBinder::DeathRecipient 427 virtual void binderDied(const wp<IBinder>& who); 428 429 private: 430 NotificationClient(const NotificationClient&); 431 NotificationClient& operator = (const NotificationClient&); 432 433 const sp<AudioFlinger> mAudioFlinger; 434 const pid_t mPid; 435 const sp<IAudioFlingerClient> mAudioFlingerClient; 436 }; 437 438 class TrackHandle; 439 class RecordHandle; 440 class RecordThread; 441 class PlaybackThread; 442 class MixerThread; 443 class DirectOutputThread; 444 class OffloadThread; 445 class DuplicatingThread; 446 class AsyncCallbackThread; 447 class Track; 448 class RecordTrack; 449 class EffectModule; 450 class EffectHandle; 451 class EffectChain; 452 struct AudioStreamOut; 453 struct AudioStreamIn; 454 455 struct stream_type_t { 456 stream_type_t() 457 : volume(1.0f), 458 mute(false) 459 { 460 } 461 float volume; 462 bool mute; 463 }; 464 465 // --- PlaybackThread --- 466 467 #include "Threads.h" 468 469 #include "Effects.h" 470 471 #include "PatchPanel.h" 472 473 // server side of the client's IAudioTrack 474 class TrackHandle : public android::BnAudioTrack { 475 public: 476 TrackHandle(const sp<PlaybackThread::Track>& track); 477 virtual ~TrackHandle(); 478 virtual sp<IMemory> getCblk() const; 479 virtual status_t start(); 480 virtual void stop(); 481 virtual void flush(); 482 virtual void pause(); 483 virtual status_t attachAuxEffect(int effectId); 484 virtual status_t allocateTimedBuffer(size_t size, 485 sp<IMemory>* buffer); 486 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 487 int64_t pts); 488 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 489 int target); 490 virtual status_t setParameters(const String8& keyValuePairs); 491 virtual status_t getTimestamp(AudioTimestamp& timestamp); 492 virtual void signal(); // signal playback thread for a change in control block 493 494 virtual status_t onTransact( 495 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 496 497 private: 498 const sp<PlaybackThread::Track> mTrack; 499 }; 500 501 // server side of the client's IAudioRecord 502 class RecordHandle : public android::BnAudioRecord { 503 public: 504 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 505 virtual ~RecordHandle(); 506 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 507 virtual void stop(); 508 virtual status_t onTransact( 509 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 510 private: 511 const sp<RecordThread::RecordTrack> mRecordTrack; 512 513 // for use from destructor 514 void stop_nonvirtual(); 515 }; 516 517 518 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 519 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 520 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 521 sp<RecordThread> openInput_l(audio_module_handle_t module, 522 audio_io_handle_t *input, 523 audio_config_t *config, 524 audio_devices_t device, 525 const String8& address, 526 audio_source_t source, 527 audio_input_flags_t flags); 528 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 529 audio_io_handle_t *output, 530 audio_config_t *config, 531 audio_devices_t devices, 532 const String8& address, 533 audio_output_flags_t flags); 534 535 void closeOutputFinish(sp<PlaybackThread> thread); 536 void closeInputFinish(sp<RecordThread> thread); 537 538 // no range check, AudioFlinger::mLock held 539 bool streamMute_l(audio_stream_type_t stream) const 540 { return mStreamTypes[stream].mute; } 541 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 542 float streamVolume_l(audio_stream_type_t stream) const 543 { return mStreamTypes[stream].volume; } 544 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 545 546 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 547 // They all share the same ID space, but the namespaces are actually independent 548 // because there are separate KeyedVectors for each kind of ID. 549 // The return value is uint32_t, but is cast to signed for some IDs. 550 // FIXME This API does not handle rollover to zero (for unsigned IDs), 551 // or from positive to negative (for signed IDs). 552 // Thus it may fail by returning an ID of the wrong sign, 553 // or by returning a non-unique ID. 554 uint32_t nextUniqueId(); 555 556 status_t moveEffectChain_l(int sessionId, 557 PlaybackThread *srcThread, 558 PlaybackThread *dstThread, 559 bool reRegister); 560 // return thread associated with primary hardware device, or NULL 561 PlaybackThread *primaryPlaybackThread_l() const; 562 audio_devices_t primaryOutputDevice_l() const; 563 564 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 565 566 567 void removeClient_l(pid_t pid); 568 void removeNotificationClient(pid_t pid); 569 bool isNonOffloadableGlobalEffectEnabled_l(); 570 void onNonOffloadableGlobalEffectEnable(); 571 572 // Store an effect chain to mOrphanEffectChains keyed vector. 573 // Called when a thread exits and effects are still attached to it. 574 // If effects are later created on the same session, they will reuse the same 575 // effect chain and same instances in the effect library. 576 // return ALREADY_EXISTS if a chain with the same session already exists in 577 // mOrphanEffectChains. Note that this should never happen as there is only one 578 // chain for a given session and it is attached to only one thread at a time. 579 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 580 // Get an effect chain for the specified session in mOrphanEffectChains and remove 581 // it if found. Returns 0 if not found (this is the most common case). 582 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 583 // Called when the last effect handle on an effect instance is removed. If this 584 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 585 // and removed from mOrphanEffectChains if it does not contain any effect. 586 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 587 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 588 589 class AudioHwDevice { 590 public: 591 enum Flags { 592 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 593 AHWD_CAN_SET_MASTER_MUTE = 0x2, 594 }; 595 596 AudioHwDevice(audio_module_handle_t handle, 597 const char *moduleName, 598 audio_hw_device_t *hwDevice, 599 Flags flags) 600 : mHandle(handle), mModuleName(strdup(moduleName)) 601 , mHwDevice(hwDevice) 602 , mFlags(flags) { } 603 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 604 605 bool canSetMasterVolume() const { 606 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 607 } 608 609 bool canSetMasterMute() const { 610 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 611 } 612 613 audio_module_handle_t handle() const { return mHandle; } 614 const char *moduleName() const { return mModuleName; } 615 audio_hw_device_t *hwDevice() const { return mHwDevice; } 616 uint32_t version() const { return mHwDevice->common.version; } 617 618 private: 619 const audio_module_handle_t mHandle; 620 const char * const mModuleName; 621 audio_hw_device_t * const mHwDevice; 622 const Flags mFlags; 623 }; 624 625 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 626 // For emphasis, we could also make all pointers to them be "const *", 627 // but that would clutter the code unnecessarily. 628 629 struct AudioStreamOut { 630 AudioHwDevice* const audioHwDev; 631 audio_stream_out_t* const stream; 632 const audio_output_flags_t flags; 633 634 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 635 636 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 637 audioHwDev(dev), stream(out), flags(flags) {} 638 }; 639 640 struct AudioStreamIn { 641 AudioHwDevice* const audioHwDev; 642 audio_stream_in_t* const stream; 643 644 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 645 646 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 647 audioHwDev(dev), stream(in) {} 648 }; 649 650 // for mAudioSessionRefs only 651 struct AudioSessionRef { 652 AudioSessionRef(int sessionid, pid_t pid) : 653 mSessionid(sessionid), mPid(pid), mCnt(1) {} 654 const int mSessionid; 655 const pid_t mPid; 656 int mCnt; 657 }; 658 659 mutable Mutex mLock; 660 // protects mClients and mNotificationClients. 661 // must be locked after mLock and ThreadBase::mLock if both must be locked 662 // avoids acquiring AudioFlinger::mLock from inside thread loop. 663 mutable Mutex mClientLock; 664 // protected by mClientLock 665 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 666 667 mutable Mutex mHardwareLock; 668 // NOTE: If both mLock and mHardwareLock mutexes must be held, 669 // always take mLock before mHardwareLock 670 671 // These two fields are immutable after onFirstRef(), so no lock needed to access 672 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 673 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 674 675 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 676 enum hardware_call_state { 677 AUDIO_HW_IDLE = 0, // no operation in progress 678 AUDIO_HW_INIT, // init_check 679 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 680 AUDIO_HW_OUTPUT_CLOSE, // unused 681 AUDIO_HW_INPUT_OPEN, // unused 682 AUDIO_HW_INPUT_CLOSE, // unused 683 AUDIO_HW_STANDBY, // unused 684 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 685 AUDIO_HW_GET_ROUTING, // unused 686 AUDIO_HW_SET_ROUTING, // unused 687 AUDIO_HW_GET_MODE, // unused 688 AUDIO_HW_SET_MODE, // set_mode 689 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 690 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 691 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 692 AUDIO_HW_SET_PARAMETER, // set_parameters 693 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 694 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 695 AUDIO_HW_GET_PARAMETER, // get_parameters 696 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 697 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 698 }; 699 700 mutable hardware_call_state mHardwareStatus; // for dump only 701 702 703 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 704 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 705 706 // member variables below are protected by mLock 707 float mMasterVolume; 708 bool mMasterMute; 709 // end of variables protected by mLock 710 711 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 712 713 // protected by mClientLock 714 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 715 716 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 717 // nextUniqueId() returns uint32_t, but this is declared int32_t 718 // because the atomic operations require an int32_t 719 720 audio_mode_t mMode; 721 bool mBtNrecIsOff; 722 723 // protected by mLock 724 Vector<AudioSessionRef*> mAudioSessionRefs; 725 726 float masterVolume_l() const; 727 bool masterMute_l() const; 728 audio_module_handle_t loadHwModule_l(const char *name); 729 730 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 731 // to be created 732 733 // Effect chains without a valid thread 734 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 735 736 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 737 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 738 private: 739 sp<Client> registerPid(pid_t pid); // always returns non-0 740 741 // for use from destructor 742 status_t closeOutput_nonvirtual(audio_io_handle_t output); 743 void closeOutputInternal_l(sp<PlaybackThread> thread); 744 status_t closeInput_nonvirtual(audio_io_handle_t input); 745 void closeInputInternal_l(sp<RecordThread> thread); 746 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 747 748 status_t checkStreamType(audio_stream_type_t stream) const; 749 750 #ifdef TEE_SINK 751 // all record threads serially share a common tee sink, which is re-created on format change 752 sp<NBAIO_Sink> mRecordTeeSink; 753 sp<NBAIO_Source> mRecordTeeSource; 754 #endif 755 756 public: 757 758 #ifdef TEE_SINK 759 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 760 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 761 762 // whether tee sink is enabled by property 763 static bool mTeeSinkInputEnabled; 764 static bool mTeeSinkOutputEnabled; 765 static bool mTeeSinkTrackEnabled; 766 767 // runtime configured size of each tee sink pipe, in frames 768 static size_t mTeeSinkInputFrames; 769 static size_t mTeeSinkOutputFrames; 770 static size_t mTeeSinkTrackFrames; 771 772 // compile-time default size of tee sink pipes, in frames 773 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 774 static const size_t kTeeSinkInputFramesDefault = 0x200000; 775 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 776 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 777 #endif 778 779 // This method reads from a variable without mLock, but the variable is updated under mLock. So 780 // we might read a stale value, or a value that's inconsistent with respect to other variables. 781 // In this case, it's safe because the return value isn't used for making an important decision. 782 // The reason we don't want to take mLock is because it could block the caller for a long time. 783 bool isLowRamDevice() const { return mIsLowRamDevice; } 784 785 private: 786 bool mIsLowRamDevice; 787 bool mIsDeviceTypeKnown; 788 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 789 790 sp<PatchPanel> mPatchPanel; 791 792 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 793 // protected by mHardwareLock 794 }; 795 796 #undef INCLUDING_FROM_AUDIOFLINGER_H 797 798 const char *formatToString(audio_format_t format); 799 800 // ---------------------------------------------------------------------------- 801 802 }; // namespace android 803 804 #endif // ANDROID_AUDIO_FLINGER_H 805
