AudioFlinger.h revision d5681bc9a38fe4cd1d591e6ae62b9c68fb851041
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 <stdint.h> 22#include <sys/types.h> 23#include <limits.h> 24 25#include <common_time/cc_helper.h> 26 27#include <media/IAudioFlinger.h> 28#include <media/IAudioFlingerClient.h> 29#include <media/IAudioTrack.h> 30#include <media/IAudioRecord.h> 31#include <media/AudioSystem.h> 32#include <media/AudioTrack.h> 33 34#include <utils/Atomic.h> 35#include <utils/Errors.h> 36#include <utils/threads.h> 37#include <utils/SortedVector.h> 38#include <utils/TypeHelpers.h> 39#include <utils/Vector.h> 40 41#include <binder/BinderService.h> 42#include <binder/MemoryDealer.h> 43 44#include <system/audio.h> 45#include <hardware/audio.h> 46#include <hardware/audio_policy.h> 47 48#include <media/AudioBufferProvider.h> 49#include <media/ExtendedAudioBufferProvider.h> 50#include "FastMixer.h" 51#include <media/nbaio/NBAIO.h> 52#include "AudioWatchdog.h" 53 54#include <powermanager/IPowerManager.h> 55 56namespace android { 57 58class audio_track_cblk_t; 59class effect_param_cblk_t; 60class AudioMixer; 61class AudioBuffer; 62class AudioResampler; 63class FastMixer; 64class ServerProxy; 65 66// ---------------------------------------------------------------------------- 67 68// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 69// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 70// Adding full support for > 2 channel capture or playback would require more than simply changing 71// this #define. There is an independent hard-coded upper limit in AudioMixer; 72// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 73// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 74// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 75#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 76 77static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 78 79#define MAX_GAIN 4096.0f 80#define MAX_GAIN_INT 0x1000 81 82#define INCLUDING_FROM_AUDIOFLINGER_H 83 84class AudioFlinger : 85 public BinderService<AudioFlinger>, 86 public BnAudioFlinger 87{ 88 friend class BinderService<AudioFlinger>; // for AudioFlinger() 89public: 90 static const char* getServiceName() { return "media.audio_flinger"; } 91 92 virtual status_t dump(int fd, const Vector<String16>& args); 93 94 // IAudioFlinger interface, in binder opcode order 95 virtual sp<IAudioTrack> createTrack( 96 audio_stream_type_t streamType, 97 uint32_t sampleRate, 98 audio_format_t format, 99 audio_channel_mask_t channelMask, 100 size_t frameCount, 101 IAudioFlinger::track_flags_t *flags, 102 const sp<IMemory>& sharedBuffer, 103 audio_io_handle_t output, 104 pid_t tid, 105 int *sessionId, 106 status_t *status); 107 108 virtual sp<IAudioRecord> openRecord( 109 audio_io_handle_t input, 110 uint32_t sampleRate, 111 audio_format_t format, 112 audio_channel_mask_t channelMask, 113 size_t frameCount, 114 IAudioFlinger::track_flags_t flags, 115 pid_t tid, 116 int *sessionId, 117 status_t *status); 118 119 virtual uint32_t sampleRate(audio_io_handle_t output) const; 120 virtual int channelCount(audio_io_handle_t output) const; 121 virtual audio_format_t format(audio_io_handle_t output) const; 122 virtual size_t frameCount(audio_io_handle_t output) const; 123 virtual uint32_t latency(audio_io_handle_t output) const; 124 125 virtual status_t setMasterVolume(float value); 126 virtual status_t setMasterMute(bool muted); 127 128 virtual float masterVolume() const; 129 virtual bool masterMute() const; 130 131 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 132 audio_io_handle_t output); 133 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 134 135 virtual float streamVolume(audio_stream_type_t stream, 136 audio_io_handle_t output) const; 137 virtual bool streamMute(audio_stream_type_t stream) const; 138 139 virtual status_t setMode(audio_mode_t mode); 140 141 virtual status_t setMicMute(bool state); 142 virtual bool getMicMute() const; 143 144 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 145 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 146 147 virtual void registerClient(const sp<IAudioFlingerClient>& client); 148 149 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 150 audio_channel_mask_t channelMask) const; 151 152 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 153 audio_devices_t *pDevices, 154 uint32_t *pSamplingRate, 155 audio_format_t *pFormat, 156 audio_channel_mask_t *pChannelMask, 157 uint32_t *pLatencyMs, 158 audio_output_flags_t flags); 159 160 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 161 audio_io_handle_t output2); 162 163 virtual status_t closeOutput(audio_io_handle_t output); 164 165 virtual status_t suspendOutput(audio_io_handle_t output); 166 167 virtual status_t restoreOutput(audio_io_handle_t output); 168 169 virtual audio_io_handle_t openInput(audio_module_handle_t module, 170 audio_devices_t *pDevices, 171 uint32_t *pSamplingRate, 172 audio_format_t *pFormat, 173 audio_channel_mask_t *pChannelMask); 174 175 virtual status_t closeInput(audio_io_handle_t input); 176 177 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 178 179 virtual status_t setVoiceVolume(float volume); 180 181 virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, 182 audio_io_handle_t output) const; 183 184 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 185 186 virtual int newAudioSessionId(); 187 188 virtual void acquireAudioSessionId(int audioSession); 189 190 virtual void releaseAudioSessionId(int audioSession); 191 192 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 193 194 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 195 196 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 197 effect_descriptor_t *descriptor) const; 198 199 virtual sp<IEffect> createEffect( 200 effect_descriptor_t *pDesc, 201 const sp<IEffectClient>& effectClient, 202 int32_t priority, 203 audio_io_handle_t io, 204 int sessionId, 205 status_t *status, 206 int *id, 207 int *enabled); 208 209 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 210 audio_io_handle_t dstOutput); 211 212 virtual audio_module_handle_t loadHwModule(const char *name); 213 214 virtual uint32_t getPrimaryOutputSamplingRate(); 215 virtual size_t getPrimaryOutputFrameCount(); 216 217 virtual status_t onTransact( 218 uint32_t code, 219 const Parcel& data, 220 Parcel* reply, 221 uint32_t flags); 222 223 // end of IAudioFlinger interface 224 225 class SyncEvent; 226 227 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 228 229 class SyncEvent : public RefBase { 230 public: 231 SyncEvent(AudioSystem::sync_event_t type, 232 int triggerSession, 233 int listenerSession, 234 sync_event_callback_t callBack, 235 void *cookie) 236 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 237 mCallback(callBack), mCookie(cookie) 238 {} 239 240 virtual ~SyncEvent() {} 241 242 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 243 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 244 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 245 AudioSystem::sync_event_t type() const { return mType; } 246 int triggerSession() const { return mTriggerSession; } 247 int listenerSession() const { return mListenerSession; } 248 void *cookie() const { return mCookie; } 249 250 private: 251 const AudioSystem::sync_event_t mType; 252 const int mTriggerSession; 253 const int mListenerSession; 254 sync_event_callback_t mCallback; 255 void * const mCookie; 256 mutable Mutex mLock; 257 }; 258 259 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 260 int triggerSession, 261 int listenerSession, 262 sync_event_callback_t callBack, 263 void *cookie); 264 265private: 266 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 267 268 audio_mode_t getMode() const { return mMode; } 269 270 bool btNrecIsOff() const { return mBtNrecIsOff; } 271 272 AudioFlinger(); 273 virtual ~AudioFlinger(); 274 275 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 276 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 277 NO_INIT : NO_ERROR; } 278 279 // RefBase 280 virtual void onFirstRef(); 281 282 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 283 audio_devices_t devices); 284 void purgeStaleEffects_l(); 285 286 // standby delay for MIXER and DUPLICATING playback threads is read from property 287 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 288 static nsecs_t mStandbyTimeInNsecs; 289 290 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 291 // AudioFlinger::setParameters() updates, other threads read w/o lock 292 static uint32_t mScreenState; 293 294 // Internal dump utilities. 295 static const int kDumpLockRetries = 50; 296 static const int kDumpLockSleepUs = 20000; 297 static bool dumpTryLock(Mutex& mutex); 298 void dumpPermissionDenial(int fd, const Vector<String16>& args); 299 void dumpClients(int fd, const Vector<String16>& args); 300 void dumpInternals(int fd, const Vector<String16>& args); 301 302 // --- Client --- 303 class Client : public RefBase { 304 public: 305 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 306 virtual ~Client(); 307 sp<MemoryDealer> heap() const; 308 pid_t pid() const { return mPid; } 309 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 310 311 bool reserveTimedTrack(); 312 void releaseTimedTrack(); 313 314 private: 315 Client(const Client&); 316 Client& operator = (const Client&); 317 const sp<AudioFlinger> mAudioFlinger; 318 const sp<MemoryDealer> mMemoryDealer; 319 const pid_t mPid; 320 321 Mutex mTimedTrackLock; 322 int mTimedTrackCount; 323 }; 324 325 // --- Notification Client --- 326 class NotificationClient : public IBinder::DeathRecipient { 327 public: 328 NotificationClient(const sp<AudioFlinger>& audioFlinger, 329 const sp<IAudioFlingerClient>& client, 330 pid_t pid); 331 virtual ~NotificationClient(); 332 333 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 334 335 // IBinder::DeathRecipient 336 virtual void binderDied(const wp<IBinder>& who); 337 338 private: 339 NotificationClient(const NotificationClient&); 340 NotificationClient& operator = (const NotificationClient&); 341 342 const sp<AudioFlinger> mAudioFlinger; 343 const pid_t mPid; 344 const sp<IAudioFlingerClient> mAudioFlingerClient; 345 }; 346 347 class TrackHandle; 348 class RecordHandle; 349 class RecordThread; 350 class PlaybackThread; 351 class MixerThread; 352 class DirectOutputThread; 353 class DuplicatingThread; 354 class Track; 355 class RecordTrack; 356 class EffectModule; 357 class EffectHandle; 358 class EffectChain; 359 struct AudioStreamOut; 360 struct AudioStreamIn; 361 362 struct stream_type_t { 363 stream_type_t() 364 : volume(1.0f), 365 mute(false) 366 { 367 } 368 float volume; 369 bool mute; 370 }; 371 372 // --- PlaybackThread --- 373 374#include "Threads.h" 375 376#include "Effects.h" 377 378 // server side of the client's IAudioTrack 379 class TrackHandle : public android::BnAudioTrack { 380 public: 381 TrackHandle(const sp<PlaybackThread::Track>& track); 382 virtual ~TrackHandle(); 383 virtual sp<IMemory> getCblk() const; 384 virtual status_t start(); 385 virtual void stop(); 386 virtual void flush(); 387 virtual void pause(); 388 virtual status_t attachAuxEffect(int effectId); 389 virtual status_t allocateTimedBuffer(size_t size, 390 sp<IMemory>* buffer); 391 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 392 int64_t pts); 393 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 394 int target); 395 virtual status_t onTransact( 396 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 397 private: 398 const sp<PlaybackThread::Track> mTrack; 399 }; 400 401 // server side of the client's IAudioRecord 402 class RecordHandle : public android::BnAudioRecord { 403 public: 404 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 405 virtual ~RecordHandle(); 406 virtual sp<IMemory> getCblk() const; 407 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 408 virtual void stop(); 409 virtual status_t onTransact( 410 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 411 private: 412 const sp<RecordThread::RecordTrack> mRecordTrack; 413 414 // for use from destructor 415 void stop_nonvirtual(); 416 }; 417 418 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 419 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 420 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 421 // no range check, AudioFlinger::mLock held 422 bool streamMute_l(audio_stream_type_t stream) const 423 { return mStreamTypes[stream].mute; } 424 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 425 float streamVolume_l(audio_stream_type_t stream) const 426 { return mStreamTypes[stream].volume; } 427 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 428 429 // allocate an audio_io_handle_t, session ID, or effect ID 430 uint32_t nextUniqueId(); 431 432 status_t moveEffectChain_l(int sessionId, 433 PlaybackThread *srcThread, 434 PlaybackThread *dstThread, 435 bool reRegister); 436 // return thread associated with primary hardware device, or NULL 437 PlaybackThread *primaryPlaybackThread_l() const; 438 audio_devices_t primaryOutputDevice_l() const; 439 440 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 441 442 443 void removeClient_l(pid_t pid); 444 void removeNotificationClient(pid_t pid); 445 446 class AudioHwDevice { 447 public: 448 enum Flags { 449 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 450 AHWD_CAN_SET_MASTER_MUTE = 0x2, 451 }; 452 453 AudioHwDevice(const char *moduleName, 454 audio_hw_device_t *hwDevice, 455 Flags flags) 456 : mModuleName(strdup(moduleName)) 457 , mHwDevice(hwDevice) 458 , mFlags(flags) { } 459 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 460 461 bool canSetMasterVolume() const { 462 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 463 } 464 465 bool canSetMasterMute() const { 466 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 467 } 468 469 const char *moduleName() const { return mModuleName; } 470 audio_hw_device_t *hwDevice() const { return mHwDevice; } 471 private: 472 const char * const mModuleName; 473 audio_hw_device_t * const mHwDevice; 474 Flags mFlags; 475 }; 476 477 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 478 // For emphasis, we could also make all pointers to them be "const *", 479 // but that would clutter the code unnecessarily. 480 481 struct AudioStreamOut { 482 AudioHwDevice* const audioHwDev; 483 audio_stream_out_t* const stream; 484 485 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 486 487 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 488 audioHwDev(dev), stream(out) {} 489 }; 490 491 struct AudioStreamIn { 492 AudioHwDevice* const audioHwDev; 493 audio_stream_in_t* const stream; 494 495 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 496 497 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 498 audioHwDev(dev), stream(in) {} 499 }; 500 501 // for mAudioSessionRefs only 502 struct AudioSessionRef { 503 AudioSessionRef(int sessionid, pid_t pid) : 504 mSessionid(sessionid), mPid(pid), mCnt(1) {} 505 const int mSessionid; 506 const pid_t mPid; 507 int mCnt; 508 }; 509 510 mutable Mutex mLock; 511 512 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 513 514 mutable Mutex mHardwareLock; 515 // NOTE: If both mLock and mHardwareLock mutexes must be held, 516 // always take mLock before mHardwareLock 517 518 // These two fields are immutable after onFirstRef(), so no lock needed to access 519 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 520 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 521 522 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 523 enum hardware_call_state { 524 AUDIO_HW_IDLE = 0, // no operation in progress 525 AUDIO_HW_INIT, // init_check 526 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 527 AUDIO_HW_OUTPUT_CLOSE, // unused 528 AUDIO_HW_INPUT_OPEN, // unused 529 AUDIO_HW_INPUT_CLOSE, // unused 530 AUDIO_HW_STANDBY, // unused 531 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 532 AUDIO_HW_GET_ROUTING, // unused 533 AUDIO_HW_SET_ROUTING, // unused 534 AUDIO_HW_GET_MODE, // unused 535 AUDIO_HW_SET_MODE, // set_mode 536 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 537 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 538 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 539 AUDIO_HW_SET_PARAMETER, // set_parameters 540 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 541 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 542 AUDIO_HW_GET_PARAMETER, // get_parameters 543 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 544 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 545 }; 546 547 mutable hardware_call_state mHardwareStatus; // for dump only 548 549 550 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 551 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 552 553 // member variables below are protected by mLock 554 float mMasterVolume; 555 bool mMasterMute; 556 // end of variables protected by mLock 557 558 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 559 560 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 561 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 562 audio_mode_t mMode; 563 bool mBtNrecIsOff; 564 565 // protected by mLock 566 Vector<AudioSessionRef*> mAudioSessionRefs; 567 568 float masterVolume_l() const; 569 bool masterMute_l() const; 570 audio_module_handle_t loadHwModule_l(const char *name); 571 572 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 573 // to be created 574 575private: 576 sp<Client> registerPid_l(pid_t pid); // always returns non-0 577 578 // for use from destructor 579 status_t closeOutput_nonvirtual(audio_io_handle_t output); 580 status_t closeInput_nonvirtual(audio_io_handle_t input); 581 582 // all record threads serially share a common tee sink, which is re-created on format change 583 sp<NBAIO_Sink> mRecordTeeSink; 584 sp<NBAIO_Source> mRecordTeeSource; 585 586public: 587 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 588}; 589 590#undef INCLUDING_FROM_AUDIOFLINGER_H 591 592// ---------------------------------------------------------------------------- 593 594}; // namespace android 595 596#endif // ANDROID_AUDIO_FLINGER_H 597