AudioPolicyManager.h revision cbd48023d0a0e3fd59955011538c0087a439f905
1/* 2 * Copyright (C) 2009 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 18#include <stdint.h> 19#include <sys/types.h> 20#include <cutils/config_utils.h> 21#include <cutils/misc.h> 22#include <utils/Timers.h> 23#include <utils/Errors.h> 24#include <utils/KeyedVector.h> 25#include <utils/SortedVector.h> 26#include "AudioPolicyInterface.h" 27 28 29namespace android { 30 31// ---------------------------------------------------------------------------- 32 33// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB 34#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 35// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB 36#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 37// Time in milliseconds during which we consider that music is still active after a music 38// track was stopped - see computeVolume() 39#define SONIFICATION_HEADSET_MUSIC_DELAY 5000 40// Time in milliseconds after media stopped playing during which we consider that the 41// sonification should be as unobtrusive as during the time media was playing. 42#define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 43// Time in milliseconds during witch some streams are muted while the audio path 44// is switched 45#define MUTE_TIME_MS 2000 46 47#define NUM_TEST_OUTPUTS 5 48 49#define NUM_VOL_CURVE_KNEES 2 50 51// Default minimum length allowed for offloading a compressed track 52// Can be overridden by the audio.offload.min.duration.secs property 53#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 54 55#define MAX_MIXER_SAMPLING_RATE 48000 56#define MAX_MIXER_CHANNEL_COUNT 2 57// See AudioPort::compareFormats() 58#define WORST_MIXER_FORMAT AUDIO_FORMAT_PCM_16_BIT 59#define BEST_MIXER_FORMAT AUDIO_FORMAT_PCM_24_BIT_PACKED 60 61// ---------------------------------------------------------------------------- 62// AudioPolicyManager implements audio policy manager behavior common to all platforms. 63// ---------------------------------------------------------------------------- 64 65class AudioPolicyManager: public AudioPolicyInterface 66#ifdef AUDIO_POLICY_TEST 67 , public Thread 68#endif //AUDIO_POLICY_TEST 69{ 70 71public: 72 AudioPolicyManager(AudioPolicyClientInterface *clientInterface); 73 virtual ~AudioPolicyManager(); 74 75 // AudioPolicyInterface 76 virtual status_t setDeviceConnectionState(audio_devices_t device, 77 audio_policy_dev_state_t state, 78 const char *device_address); 79 virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 80 const char *device_address); 81 virtual void setPhoneState(audio_mode_t state); 82 virtual void setForceUse(audio_policy_force_use_t usage, 83 audio_policy_forced_cfg_t config); 84 virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 85 virtual void setSystemProperty(const char* property, const char* value); 86 virtual status_t initCheck(); 87 virtual audio_io_handle_t getOutput(audio_stream_type_t stream, 88 uint32_t samplingRate, 89 audio_format_t format, 90 audio_channel_mask_t channelMask, 91 audio_output_flags_t flags, 92 const audio_offload_info_t *offloadInfo); 93 virtual audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr, 94 uint32_t samplingRate, 95 audio_format_t format, 96 audio_channel_mask_t channelMask, 97 audio_output_flags_t flags, 98 const audio_offload_info_t *offloadInfo); 99 virtual status_t startOutput(audio_io_handle_t output, 100 audio_stream_type_t stream, 101 int session = 0); 102 virtual status_t stopOutput(audio_io_handle_t output, 103 audio_stream_type_t stream, 104 int session = 0); 105 virtual void releaseOutput(audio_io_handle_t output); 106 virtual audio_io_handle_t getInput(audio_source_t inputSource, 107 uint32_t samplingRate, 108 audio_format_t format, 109 audio_channel_mask_t channelMask, 110 audio_in_acoustics_t acoustics, 111 audio_input_flags_t flags); 112 113 // indicates to the audio policy manager that the input starts being used. 114 virtual status_t startInput(audio_io_handle_t input); 115 116 // indicates to the audio policy manager that the input stops being used. 117 virtual status_t stopInput(audio_io_handle_t input); 118 virtual void releaseInput(audio_io_handle_t input); 119 virtual void closeAllInputs(); 120 virtual void initStreamVolume(audio_stream_type_t stream, 121 int indexMin, 122 int indexMax); 123 virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, 124 int index, 125 audio_devices_t device); 126 virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, 127 int *index, 128 audio_devices_t device); 129 130 // return the strategy corresponding to a given stream type 131 virtual uint32_t getStrategyForStream(audio_stream_type_t stream); 132 // return the strategy corresponding to the given audio attributes 133 virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr); 134 135 // return the enabled output devices for the given stream type 136 virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); 137 138 virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); 139 virtual status_t registerEffect(const effect_descriptor_t *desc, 140 audio_io_handle_t io, 141 uint32_t strategy, 142 int session, 143 int id); 144 virtual status_t unregisterEffect(int id); 145 virtual status_t setEffectEnabled(int id, bool enabled); 146 147 virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; 148 // return whether a stream is playing remotely, override to change the definition of 149 // local/remote playback, used for instance by notification manager to not make 150 // media players lose audio focus when not playing locally 151 virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; 152 virtual bool isSourceActive(audio_source_t source) const; 153 154 virtual status_t dump(int fd); 155 156 virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); 157 158 virtual status_t listAudioPorts(audio_port_role_t role, 159 audio_port_type_t type, 160 unsigned int *num_ports, 161 struct audio_port *ports, 162 unsigned int *generation); 163 virtual status_t getAudioPort(struct audio_port *port); 164 virtual status_t createAudioPatch(const struct audio_patch *patch, 165 audio_patch_handle_t *handle, 166 uid_t uid); 167 virtual status_t releaseAudioPatch(audio_patch_handle_t handle, 168 uid_t uid); 169 virtual status_t listAudioPatches(unsigned int *num_patches, 170 struct audio_patch *patches, 171 unsigned int *generation); 172 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 173 virtual void clearAudioPatches(uid_t uid); 174 175protected: 176 177 enum routing_strategy { 178 STRATEGY_MEDIA, 179 STRATEGY_PHONE, 180 STRATEGY_SONIFICATION, 181 STRATEGY_SONIFICATION_RESPECTFUL, 182 STRATEGY_DTMF, 183 STRATEGY_ENFORCED_AUDIBLE, 184 NUM_STRATEGIES 185 }; 186 187 // 4 points to define the volume attenuation curve, each characterized by the volume 188 // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. 189 // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() 190 191 enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; 192 193 class VolumeCurvePoint 194 { 195 public: 196 int mIndex; 197 float mDBAttenuation; 198 }; 199 200 // device categories used for volume curve management. 201 enum device_category { 202 DEVICE_CATEGORY_HEADSET, 203 DEVICE_CATEGORY_SPEAKER, 204 DEVICE_CATEGORY_EARPIECE, 205 DEVICE_CATEGORY_CNT 206 }; 207 208 class HwModule; 209 210 class AudioGain: public RefBase 211 { 212 public: 213 AudioGain(int index, bool useInChannelMask); 214 virtual ~AudioGain() {} 215 216 void dump(int fd, int spaces, int index) const; 217 218 void getDefaultConfig(struct audio_gain_config *config); 219 status_t checkConfig(const struct audio_gain_config *config); 220 int mIndex; 221 struct audio_gain mGain; 222 bool mUseInChannelMask; 223 }; 224 225 class AudioPort: public virtual RefBase 226 { 227 public: 228 AudioPort(const String8& name, audio_port_type_t type, 229 audio_port_role_t role, const sp<HwModule>& module); 230 virtual ~AudioPort() {} 231 232 virtual void toAudioPort(struct audio_port *port) const; 233 234 void loadSamplingRates(char *name); 235 void loadFormats(char *name); 236 void loadOutChannels(char *name); 237 void loadInChannels(char *name); 238 239 audio_gain_mode_t loadGainMode(char *name); 240 void loadGain(cnode *root, int index); 241 void loadGains(cnode *root); 242 243 // searches for an exact match 244 status_t checkExactSamplingRate(uint32_t samplingRate) const; 245 // searches for a compatible match, and returns the best match via updatedSamplingRate 246 status_t checkCompatibleSamplingRate(uint32_t samplingRate, 247 uint32_t *updatedSamplingRate) const; 248 // searches for an exact match 249 status_t checkExactChannelMask(audio_channel_mask_t channelMask) const; 250 // searches for a compatible match, currently implemented for input channel masks only 251 status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const; 252 status_t checkFormat(audio_format_t format) const; 253 status_t checkGain(const struct audio_gain_config *gainConfig, int index) const; 254 255 uint32_t pickSamplingRate() const; 256 audio_channel_mask_t pickChannelMask() const; 257 audio_format_t pickFormat() const; 258 259 static const audio_format_t sPcmFormatCompareTable[]; 260 static int compareFormats(audio_format_t format1, audio_format_t format2); 261 262 void dump(int fd, int spaces) const; 263 264 String8 mName; 265 audio_port_type_t mType; 266 audio_port_role_t mRole; 267 bool mUseInChannelMask; 268 // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats 269 // indicates the supported parameters should be read from the output stream 270 // after it is opened for the first time 271 Vector <uint32_t> mSamplingRates; // supported sampling rates 272 Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks 273 Vector <audio_format_t> mFormats; // supported audio formats 274 Vector < sp<AudioGain> > mGains; // gain controllers 275 sp<HwModule> mModule; // audio HW module exposing this I/O stream 276 audio_output_flags_t mFlags; // attribute flags (e.g primary output, 277 // direct output...). For outputs only. 278 }; 279 280 class AudioPortConfig: public virtual RefBase 281 { 282 public: 283 AudioPortConfig(); 284 virtual ~AudioPortConfig() {} 285 286 status_t applyAudioPortConfig(const struct audio_port_config *config, 287 struct audio_port_config *backupConfig = NULL); 288 virtual void toAudioPortConfig(struct audio_port_config *dstConfig, 289 const struct audio_port_config *srcConfig = NULL) const = 0; 290 sp<AudioPort> mAudioPort; 291 uint32_t mSamplingRate; 292 audio_format_t mFormat; 293 audio_channel_mask_t mChannelMask; 294 struct audio_gain_config mGain; 295 }; 296 297 298 class AudioPatch: public RefBase 299 { 300 public: 301 AudioPatch(audio_patch_handle_t handle, 302 const struct audio_patch *patch, uid_t uid) : 303 mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {} 304 305 audio_patch_handle_t mHandle; 306 struct audio_patch mPatch; 307 uid_t mUid; 308 audio_patch_handle_t mAfPatchHandle; 309 }; 310 311 class DeviceDescriptor: public AudioPort, public AudioPortConfig 312 { 313 public: 314 DeviceDescriptor(const String8& name, audio_devices_t type); 315 316 virtual ~DeviceDescriptor() {} 317 318 bool equals(const sp<DeviceDescriptor>& other) const; 319 virtual void toAudioPortConfig(struct audio_port_config *dstConfig, 320 const struct audio_port_config *srcConfig = NULL) const; 321 322 virtual void toAudioPort(struct audio_port *port) const; 323 324 status_t dump(int fd, int spaces, int index) const; 325 326 audio_devices_t mDeviceType; 327 String8 mAddress; 328 audio_port_handle_t mId; 329 }; 330 331 class DeviceVector : public SortedVector< sp<DeviceDescriptor> > 332 { 333 public: 334 DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} 335 336 ssize_t add(const sp<DeviceDescriptor>& item); 337 ssize_t remove(const sp<DeviceDescriptor>& item); 338 ssize_t indexOf(const sp<DeviceDescriptor>& item) const; 339 340 audio_devices_t types() const { return mDeviceTypes; } 341 342 void loadDevicesFromType(audio_devices_t types); 343 void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); 344 345 sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const; 346 DeviceVector getDevicesFromType(audio_devices_t types) const; 347 sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const; 348 sp<DeviceDescriptor> getDeviceFromName(const String8& name) const; 349 DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address) 350 const; 351 352 private: 353 void refreshTypes(); 354 audio_devices_t mDeviceTypes; 355 }; 356 357 // the IOProfile class describes the capabilities of an output or input stream. 358 // It is currently assumed that all combination of listed parameters are supported. 359 // It is used by the policy manager to determine if an output or input is suitable for 360 // a given use case, open/close it accordingly and connect/disconnect audio tracks 361 // to/from it. 362 class IOProfile : public AudioPort 363 { 364 public: 365 IOProfile(const String8& name, audio_port_role_t role, const sp<HwModule>& module); 366 virtual ~IOProfile(); 367 368 // This method is used for both output and input. 369 // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate. 370 // For input, flags is interpreted as audio_input_flags_t. 371 // TODO: merge audio_output_flags_t and audio_input_flags_t. 372 bool isCompatibleProfile(audio_devices_t device, 373 uint32_t samplingRate, 374 uint32_t *updatedSamplingRate, 375 audio_format_t format, 376 audio_channel_mask_t channelMask, 377 audio_output_flags_t flags) const; 378 379 void dump(int fd); 380 void log(); 381 382 DeviceVector mSupportedDevices; // supported devices 383 // (devices this output can be routed to) 384 }; 385 386 class HwModule : public RefBase 387 { 388 public: 389 HwModule(const char *name); 390 ~HwModule(); 391 392 status_t loadOutput(cnode *root); 393 status_t loadInput(cnode *root); 394 status_t loadDevice(cnode *root); 395 396 void dump(int fd); 397 398 const char *const mName; // base name of the audio HW module (primary, a2dp ...) 399 uint32_t mHalVersion; // audio HAL API version 400 audio_module_handle_t mHandle; 401 Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module 402 Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module 403 DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf 404 405 }; 406 407 // default volume curve 408 static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT]; 409 // default volume curve for media strategy 410 static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT]; 411 // volume curve for media strategy on speakers 412 static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT]; 413 static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT]; 414 // volume curve for sonification strategy on speakers 415 static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT]; 416 static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT]; 417 static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT]; 418 static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT]; 419 static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT]; 420 static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; 421 static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; 422 // default volume curves per stream and device category. See initializeVolumeCurves() 423 static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; 424 425 // descriptor for audio outputs. Used to maintain current configuration of each opened audio output 426 // and keep track of the usage of this output by each audio stream type. 427 class AudioOutputDescriptor: public AudioPortConfig 428 { 429 public: 430 AudioOutputDescriptor(const sp<IOProfile>& profile); 431 432 status_t dump(int fd); 433 434 audio_devices_t device() const; 435 void changeRefCount(audio_stream_type_t stream, int delta); 436 437 bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } 438 audio_devices_t supportedDevices(); 439 uint32_t latency(); 440 bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); 441 bool isActive(uint32_t inPastMs = 0) const; 442 bool isStreamActive(audio_stream_type_t stream, 443 uint32_t inPastMs = 0, 444 nsecs_t sysTime = 0) const; 445 bool isStrategyActive(routing_strategy strategy, 446 uint32_t inPastMs = 0, 447 nsecs_t sysTime = 0) const; 448 449 virtual void toAudioPortConfig(struct audio_port_config *dstConfig, 450 const struct audio_port_config *srcConfig = NULL) const; 451 void toAudioPort(struct audio_port *port) const; 452 453 audio_port_handle_t mId; 454 audio_io_handle_t mIoHandle; // output handle 455 uint32_t mLatency; // 456 audio_output_flags_t mFlags; // 457 audio_devices_t mDevice; // current device this output is routed to 458 audio_patch_handle_t mPatchHandle; 459 uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output 460 nsecs_t mStopTime[AUDIO_STREAM_CNT]; 461 sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output 462 sp<AudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output 463 float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume 464 int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter 465 const sp<IOProfile> mProfile; // I/O profile this output derives from 466 bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible 467 // device selection. See checkDeviceMuteStrategies() 468 uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) 469 }; 470 471 // descriptor for audio inputs. Used to maintain current configuration of each opened audio input 472 // and keep track of the usage of this input. 473 class AudioInputDescriptor: public AudioPortConfig 474 { 475 public: 476 AudioInputDescriptor(const sp<IOProfile>& profile); 477 478 status_t dump(int fd); 479 480 audio_port_handle_t mId; 481 audio_io_handle_t mIoHandle; // input handle 482 audio_devices_t mDevice; // current device this input is routed to 483 audio_patch_handle_t mPatchHandle; 484 uint32_t mRefCount; // number of AudioRecord clients using this output 485 uint32_t mOpenRefCount; 486 audio_source_t mInputSource; // input source selected by application (mediarecorder.h) 487 const sp<IOProfile> mProfile; // I/O profile this output derives from 488 489 virtual void toAudioPortConfig(struct audio_port_config *dstConfig, 490 const struct audio_port_config *srcConfig = NULL) const; 491 void toAudioPort(struct audio_port *port) const; 492 }; 493 494 // stream descriptor used for volume control 495 class StreamDescriptor 496 { 497 public: 498 StreamDescriptor(); 499 500 int getVolumeIndex(audio_devices_t device); 501 void dump(int fd); 502 503 int mIndexMin; // min volume index 504 int mIndexMax; // max volume index 505 KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device 506 bool mCanBeMuted; // true is the stream can be muted 507 508 const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; 509 }; 510 511 // stream descriptor used for volume control 512 class EffectDescriptor : public RefBase 513 { 514 public: 515 516 status_t dump(int fd); 517 518 int mIo; // io the effect is attached to 519 routing_strategy mStrategy; // routing strategy the effect is associated to 520 int mSession; // audio session the effect is on 521 effect_descriptor_t mDesc; // effect descriptor 522 bool mEnabled; // enabled state: CPU load being used or not 523 }; 524 525 void addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc); 526 void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc); 527 528 // return the strategy corresponding to a given stream type 529 static routing_strategy getStrategy(audio_stream_type_t stream); 530 531 // return appropriate device for streams handled by the specified strategy according to current 532 // phone state, connected devices... 533 // if fromCache is true, the device is returned from mDeviceForStrategy[], 534 // otherwise it is determine by current state 535 // (device connected,phone state, force use, a2dp output...) 536 // This allows to: 537 // 1 speed up process when the state is stable (when starting or stopping an output) 538 // 2 access to either current device selection (fromCache == true) or 539 // "future" device selection (fromCache == false) when called from a context 540 // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND 541 // before updateDevicesAndOutputs() is called. 542 virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, 543 bool fromCache); 544 545 // change the route of the specified output. Returns the number of ms we have slept to 546 // allow new routing to take effect in certain cases. 547 uint32_t setOutputDevice(audio_io_handle_t output, 548 audio_devices_t device, 549 bool force = false, 550 int delayMs = 0, 551 audio_patch_handle_t *patchHandle = NULL, 552 const char* address = NULL); 553 status_t resetOutputDevice(audio_io_handle_t output, 554 int delayMs = 0, 555 audio_patch_handle_t *patchHandle = NULL); 556 status_t setInputDevice(audio_io_handle_t input, 557 audio_devices_t device, 558 bool force = false, 559 audio_patch_handle_t *patchHandle = NULL); 560 status_t resetInputDevice(audio_io_handle_t input, 561 audio_patch_handle_t *patchHandle = NULL); 562 563 // select input device corresponding to requested audio source 564 virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource); 565 566 // return io handle of active input or 0 if no input is active 567 // Only considers inputs from physical devices (e.g. main mic, headset mic) when 568 // ignoreVirtualInputs is true. 569 audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); 570 571 // initialize volume curves for each strategy and device category 572 void initializeVolumeCurves(); 573 574 // compute the actual volume for a given stream according to the requested index and a particular 575 // device 576 virtual float computeVolume(audio_stream_type_t stream, int index, 577 audio_io_handle_t output, audio_devices_t device); 578 579 // check that volume change is permitted, compute and send new volume to audio hardware 580 status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, 581 audio_devices_t device, int delayMs = 0, bool force = false); 582 583 // apply all stream volumes to the specified output and device 584 void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); 585 586 // Mute or unmute all streams handled by the specified strategy on the specified output 587 void setStrategyMute(routing_strategy strategy, 588 bool on, 589 audio_io_handle_t output, 590 int delayMs = 0, 591 audio_devices_t device = (audio_devices_t)0); 592 593 // Mute or unmute the stream on the specified output 594 void setStreamMute(audio_stream_type_t stream, 595 bool on, 596 audio_io_handle_t output, 597 int delayMs = 0, 598 audio_devices_t device = (audio_devices_t)0); 599 600 // handle special cases for sonification strategy while in call: mute streams or replace by 601 // a special tone in the device used for communication 602 void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange); 603 604 // true if device is in a telephony or VoIP call 605 virtual bool isInCall(); 606 607 // true if given state represents a device in a telephony or VoIP call 608 virtual bool isStateInCall(int state); 609 610 // when a device is connected, checks if an open output can be routed 611 // to this device. If none is open, tries to open one of the available outputs. 612 // Returns an output suitable to this device or 0. 613 // when a device is disconnected, checks if an output is not used any more and 614 // returns its handle if any. 615 // transfers the audio tracks and effects from one output thread to another accordingly. 616 status_t checkOutputsForDevice(audio_devices_t device, 617 audio_policy_dev_state_t state, 618 SortedVector<audio_io_handle_t>& outputs, 619 const String8 address); 620 621 status_t checkInputsForDevice(audio_devices_t device, 622 audio_policy_dev_state_t state, 623 SortedVector<audio_io_handle_t>& inputs, 624 const String8 address); 625 626 // close an output and its companion duplicating output. 627 void closeOutput(audio_io_handle_t output); 628 629 // checks and if necessary changes outputs used for all strategies. 630 // must be called every time a condition that affects the output choice for a given strategy 631 // changes: connected device, phone state, force use... 632 // Must be called before updateDevicesAndOutputs() 633 void checkOutputForStrategy(routing_strategy strategy); 634 635 // Same as checkOutputForStrategy() but for a all strategies in order of priority 636 void checkOutputForAllStrategies(); 637 638 // manages A2DP output suspend/restore according to phone state and BT SCO usage 639 void checkA2dpSuspend(); 640 641 // returns the A2DP output handle if it is open or 0 otherwise 642 audio_io_handle_t getA2dpOutput(); 643 644 // selects the most appropriate device on output for current state 645 // must be called every time a condition that affects the device choice for a given output is 646 // changed: connected device, phone state, force use, output start, output stop.. 647 // see getDeviceForStrategy() for the use of fromCache parameter 648 audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); 649 650 // updates cache of device used by all strategies (mDeviceForStrategy[]) 651 // must be called every time a condition that affects the device choice for a given strategy is 652 // changed: connected device, phone state, force use... 653 // cached values are used by getDeviceForStrategy() if parameter fromCache is true. 654 // Must be called after checkOutputForAllStrategies() 655 void updateDevicesAndOutputs(); 656 657 // selects the most appropriate device on input for current state 658 audio_devices_t getNewInputDevice(audio_io_handle_t input); 659 660 virtual uint32_t getMaxEffectsCpuLoad(); 661 virtual uint32_t getMaxEffectsMemory(); 662#ifdef AUDIO_POLICY_TEST 663 virtual bool threadLoop(); 664 void exit(); 665 int testOutputIndex(audio_io_handle_t output); 666#endif //AUDIO_POLICY_TEST 667 668 status_t setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled); 669 670 // returns the category the device belongs to with regard to volume curve management 671 static device_category getDeviceCategory(audio_devices_t device); 672 673 // extract one device relevant for volume control from multiple device selection 674 static audio_devices_t getDeviceForVolume(audio_devices_t device); 675 676 SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, 677 DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs); 678 bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, 679 SortedVector<audio_io_handle_t>& outputs2); 680 681 // mute/unmute strategies using an incompatible device combination 682 // if muting, wait for the audio in pcm buffer to be drained before proceeding 683 // if unmuting, unmute only after the specified delay 684 // Returns the number of ms waited 685 uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc, 686 audio_devices_t prevDevice, 687 uint32_t delayMs); 688 689 audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs, 690 audio_output_flags_t flags); 691 // samplingRate parameter is an in/out and so may be modified 692 sp<IOProfile> getInputProfile(audio_devices_t device, 693 uint32_t& samplingRate, 694 audio_format_t format, 695 audio_channel_mask_t channelMask, 696 audio_input_flags_t flags); 697 sp<IOProfile> getProfileForDirectOutput(audio_devices_t device, 698 uint32_t samplingRate, 699 audio_format_t format, 700 audio_channel_mask_t channelMask, 701 audio_output_flags_t flags); 702 703 audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs); 704 705 bool isNonOffloadableEffectEnabled(); 706 707 status_t addAudioPatch(audio_patch_handle_t handle, 708 const sp<AudioPatch>& patch); 709 status_t removeAudioPatch(audio_patch_handle_t handle); 710 711 sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const; 712 sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const; 713 sp<HwModule> getModuleForDevice(audio_devices_t device) const; 714 sp<HwModule> getModuleFromName(const char *name) const; 715 // 716 // Audio policy configuration file parsing (audio_policy.conf) 717 // 718 static uint32_t stringToEnum(const struct StringToEnum *table, 719 size_t size, 720 const char *name); 721 static const char *enumToString(const struct StringToEnum *table, 722 size_t size, 723 uint32_t value); 724 static bool stringToBool(const char *value); 725 static audio_output_flags_t parseFlagNames(char *name); 726 static audio_devices_t parseDeviceNames(char *name); 727 void loadHwModule(cnode *root); 728 void loadHwModules(cnode *root); 729 void loadGlobalConfig(cnode *root, const sp<HwModule>& module); 730 status_t loadAudioPolicyConfig(const char *path); 731 void defaultAudioPolicyConfig(void); 732 733 734 uid_t mUidCached; 735 AudioPolicyClientInterface *mpClientInterface; // audio policy client interface 736 audio_io_handle_t mPrimaryOutput; // primary output handle 737 // list of descriptors for outputs currently opened 738 DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs; 739 // copy of mOutputs before setDeviceConnectionState() opens new outputs 740 // reset to mOutputs when updateDevicesAndOutputs() is called. 741 DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs; 742 DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > mInputs; // list of input descriptors 743 DeviceVector mAvailableOutputDevices; // all available output devices 744 DeviceVector mAvailableInputDevices; // all available input devices 745 int mPhoneState; // current phone state 746 audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration 747 748 StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control 749 bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected 750 audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; 751 float mLastVoiceVolume; // last voice volume value sent to audio HAL 752 753 // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units 754 static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; 755 // Maximum memory allocated to audio effects in KB 756 static const uint32_t MAX_EFFECTS_MEMORY = 512; 757 uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects 758 uint32_t mTotalEffectsMemory; // current memory used by effects 759 KeyedVector<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects 760 bool mA2dpSuspended; // true if A2DP output is suspended 761 sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time 762 bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path 763 // to boost soft sounds, used to adjust volume curves accordingly 764 765 Vector < sp<HwModule> > mHwModules; 766 volatile int32_t mNextUniqueId; 767 volatile int32_t mAudioPortGeneration; 768 769 DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches; 770 771#ifdef AUDIO_POLICY_TEST 772 Mutex mLock; 773 Condition mWaitWorkCV; 774 775 int mCurOutput; 776 bool mDirectOutput; 777 audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; 778 int mTestInput; 779 uint32_t mTestDevice; 780 uint32_t mTestSamplingRate; 781 uint32_t mTestFormat; 782 uint32_t mTestChannels; 783 uint32_t mTestLatencyMs; 784#endif //AUDIO_POLICY_TEST 785 786private: 787 static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, 788 int indexInUi); 789 // updates device caching and output for streams that can influence the 790 // routing of notifications 791 void handleNotificationRoutingForStream(audio_stream_type_t stream); 792 static bool isVirtualInputDevice(audio_devices_t device); 793 static bool deviceDistinguishesOnAddress(audio_devices_t device); 794 // find the outputs on a given output descriptor that have the given address. 795 // to be called on an AudioOutputDescriptor whose supported devices (as defined 796 // in mProfile->mSupportedDevices) matches the device whose address is to be matched. 797 // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one 798 // where addresses are used to distinguish between one connected device and another. 799 void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/, 800 const String8 address /*in*/, 801 SortedVector<audio_io_handle_t>& outputs /*out*/); 802 uint32_t nextUniqueId(); 803 uint32_t nextAudioPortGeneration(); 804 uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; } 805 // converts device address to string sent to audio HAL via setParameters 806 static String8 addressToParameter(audio_devices_t device, const String8 address); 807 // internal method to return the output handle for the given device and format 808 audio_io_handle_t getOutputForDevice( 809 audio_devices_t device, 810 audio_stream_type_t stream, 811 uint32_t samplingRate, 812 audio_format_t format, 813 audio_channel_mask_t channelMask, 814 audio_output_flags_t flags, 815 const audio_offload_info_t *offloadInfo); 816 // internal function to derive a stream type value from audio attributes 817 audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr); 818}; 819 820}; 821