Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 1 | /* |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 2 | * Copyright (c) 2014, The Linux Foundation. All rights reserved. |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 3 | * Not a contribution. |
| 4 | * |
| 5 | * Copyright (C) 2009 The Android Open Source Project |
| 6 | * |
| 7 | * Licensed under the Apache License, Version 2.0 (the "License"); |
| 8 | * you may not use this file except in compliance with the License. |
| 9 | * You may obtain a copy of the License at |
| 10 | * |
| 11 | * http://www.apache.org/licenses/LICENSE-2.0 |
| 12 | * |
| 13 | * Unless required by applicable law or agreed to in writing, software |
| 14 | * distributed under the License is distributed on an "AS IS" BASIS, |
| 15 | * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| 16 | * See the License for the specific language governing permissions and |
| 17 | * limitations under the License. |
| 18 | */ |
| 19 | |
| 20 | |
| 21 | #include <stdint.h> |
| 22 | #include <sys/types.h> |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 23 | #include <cutils/config_utils.h> |
| 24 | #include <cutils/misc.h> |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 25 | #include <utils/Timers.h> |
| 26 | #include <utils/Errors.h> |
| 27 | #include <utils/KeyedVector.h> |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 28 | #include <utils/SortedVector.h> |
| 29 | #include "AudioPolicyInterface.h" |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 30 | |
| 31 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 32 | namespace android { |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 33 | |
| 34 | // ---------------------------------------------------------------------------- |
| 35 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 36 | // Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB |
| 37 | #define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 |
| 38 | // Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB |
| 39 | #define SONIFICATION_HEADSET_VOLUME_MIN 0.016 |
| 40 | // Time in milliseconds during which we consider that music is still active after a music |
| 41 | // track was stopped - see computeVolume() |
| 42 | #define SONIFICATION_HEADSET_MUSIC_DELAY 5000 |
| 43 | // Time in milliseconds after media stopped playing during which we consider that the |
| 44 | // sonification should be as unobtrusive as during the time media was playing. |
| 45 | #define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 |
| 46 | // Time in milliseconds during witch some streams are muted while the audio path |
| 47 | // is switched |
| 48 | #define MUTE_TIME_MS 2000 |
| 49 | |
| 50 | #define NUM_TEST_OUTPUTS 5 |
| 51 | |
| 52 | #define NUM_VOL_CURVE_KNEES 2 |
| 53 | |
| 54 | // Default minimum length allowed for offloading a compressed track |
| 55 | // Can be overridden by the audio.offload.min.duration.secs property |
| 56 | #define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 |
| 57 | |
| 58 | // ---------------------------------------------------------------------------- |
| 59 | // AudioPolicyManager implements audio policy manager behavior common to all platforms. |
| 60 | // ---------------------------------------------------------------------------- |
| 61 | |
| 62 | class AudioPolicyManager: public AudioPolicyInterface |
| 63 | #ifdef AUDIO_POLICY_TEST |
| 64 | , public Thread |
| 65 | #endif //AUDIO_POLICY_TEST |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 66 | { |
| 67 | |
| 68 | public: |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 69 | AudioPolicyManager(AudioPolicyClientInterface *clientInterface); |
| 70 | virtual ~AudioPolicyManager(); |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 71 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 72 | // AudioPolicyInterface |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 73 | virtual status_t setDeviceConnectionState(audio_devices_t device, |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 74 | audio_policy_dev_state_t state, |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 75 | const char *device_address); |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 76 | virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, |
| 77 | const char *device_address); |
| 78 | virtual void setPhoneState(audio_mode_t state); |
| 79 | virtual void setForceUse(audio_policy_force_use_t usage, |
| 80 | audio_policy_forced_cfg_t config); |
| 81 | virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); |
| 82 | virtual void setSystemProperty(const char* property, const char* value); |
| 83 | virtual status_t initCheck(); |
| 84 | virtual audio_io_handle_t getOutput(audio_stream_type_t stream, |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 85 | uint32_t samplingRate, |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 86 | audio_format_t format, |
| 87 | audio_channel_mask_t channelMask, |
| 88 | audio_output_flags_t flags, |
| 89 | const audio_offload_info_t *offloadInfo); |
| 90 | virtual status_t startOutput(audio_io_handle_t output, |
| 91 | audio_stream_type_t stream, |
| 92 | int session = 0); |
| 93 | virtual status_t stopOutput(audio_io_handle_t output, |
| 94 | audio_stream_type_t stream, |
| 95 | int session = 0); |
| 96 | virtual void releaseOutput(audio_io_handle_t output); |
| 97 | virtual audio_io_handle_t getInput(audio_source_t inputSource, |
| 98 | uint32_t samplingRate, |
| 99 | audio_format_t format, |
| 100 | audio_channel_mask_t channelMask, |
| 101 | audio_in_acoustics_t acoustics); |
| 102 | |
| 103 | // indicates to the audio policy manager that the input starts being used. |
| 104 | virtual status_t startInput(audio_io_handle_t input); |
| 105 | |
| 106 | // indicates to the audio policy manager that the input stops being used. |
| 107 | virtual status_t stopInput(audio_io_handle_t input); |
| 108 | virtual void releaseInput(audio_io_handle_t input); |
| 109 | virtual void closeAllInputs(); |
| 110 | virtual void initStreamVolume(audio_stream_type_t stream, |
| 111 | int indexMin, |
| 112 | int indexMax); |
| 113 | virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, |
| 114 | int index, |
| 115 | audio_devices_t device); |
| 116 | virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, |
| 117 | int *index, |
| 118 | audio_devices_t device); |
| 119 | |
| 120 | // return the strategy corresponding to a given stream type |
| 121 | virtual uint32_t getStrategyForStream(audio_stream_type_t stream); |
| 122 | |
| 123 | // return the enabled output devices for the given stream type |
| 124 | virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); |
| 125 | |
| 126 | virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); |
| 127 | virtual status_t registerEffect(const effect_descriptor_t *desc, |
| 128 | audio_io_handle_t io, |
| 129 | uint32_t strategy, |
| 130 | int session, |
| 131 | int id); |
| 132 | virtual status_t unregisterEffect(int id); |
| 133 | virtual status_t setEffectEnabled(int id, bool enabled); |
| 134 | |
| 135 | virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; |
| 136 | // return whether a stream is playing remotely, override to change the definition of |
| 137 | // local/remote playback, used for instance by notification manager to not make |
| 138 | // media players lose audio focus when not playing locally |
| 139 | virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; |
| 140 | virtual bool isSourceActive(audio_source_t source) const; |
| 141 | |
| 142 | virtual status_t dump(int fd); |
Naresh Tanniru | 36c0893 | 2014-01-27 18:40:53 +0530 | [diff] [blame] | 143 | |
| 144 | virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); |
| 145 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 146 | virtual status_t listAudioPorts(audio_port_role_t role, |
| 147 | audio_port_type_t type, |
| 148 | unsigned int *num_ports, |
| 149 | struct audio_port *ports, |
| 150 | unsigned int *generation); |
| 151 | virtual status_t getAudioPort(struct audio_port *port); |
| 152 | virtual status_t createAudioPatch(const struct audio_patch *patch, |
| 153 | audio_patch_handle_t *handle, |
| 154 | uid_t uid); |
| 155 | virtual status_t releaseAudioPatch(audio_patch_handle_t handle, |
| 156 | uid_t uid); |
| 157 | virtual status_t listAudioPatches(unsigned int *num_patches, |
| 158 | struct audio_patch *patches, |
| 159 | unsigned int *generation); |
| 160 | virtual status_t setAudioPortConfig(const struct audio_port_config *config); |
| 161 | virtual void clearAudioPatches(uid_t uid); |
Karthik Reddy Katta | 060a6c4 | 2014-05-20 15:21:28 +0530 | [diff] [blame] | 162 | |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 163 | protected: |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 164 | |
| 165 | enum routing_strategy { |
| 166 | STRATEGY_MEDIA, |
| 167 | STRATEGY_PHONE, |
| 168 | STRATEGY_SONIFICATION, |
| 169 | STRATEGY_SONIFICATION_RESPECTFUL, |
| 170 | STRATEGY_DTMF, |
| 171 | STRATEGY_ENFORCED_AUDIBLE, |
| 172 | NUM_STRATEGIES |
| 173 | }; |
| 174 | |
| 175 | // 4 points to define the volume attenuation curve, each characterized by the volume |
| 176 | // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. |
| 177 | // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() |
| 178 | |
| 179 | enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; |
| 180 | |
| 181 | class VolumeCurvePoint |
| 182 | { |
| 183 | public: |
| 184 | int mIndex; |
| 185 | float mDBAttenuation; |
| 186 | }; |
| 187 | |
| 188 | // device categories used for volume curve management. |
| 189 | enum device_category { |
| 190 | DEVICE_CATEGORY_HEADSET, |
| 191 | DEVICE_CATEGORY_SPEAKER, |
| 192 | DEVICE_CATEGORY_EARPIECE, |
| 193 | DEVICE_CATEGORY_CNT |
| 194 | }; |
| 195 | |
| 196 | class HwModule; |
| 197 | |
| 198 | class AudioGain: public RefBase |
| 199 | { |
| 200 | public: |
| 201 | AudioGain(); |
| 202 | virtual ~AudioGain() {} |
| 203 | |
| 204 | void dump(int fd, int spaces, int index) const; |
| 205 | |
| 206 | struct audio_gain mGain; |
| 207 | }; |
| 208 | |
| 209 | class AudioPort: public RefBase |
| 210 | { |
| 211 | public: |
| 212 | AudioPort(const String8& name, audio_port_type_t type, |
| 213 | audio_port_role_t role, HwModule *module) : |
| 214 | mName(name), mType(type), mRole(role), mModule(module) {} |
| 215 | virtual ~AudioPort() {} |
| 216 | |
| 217 | virtual void toAudioPort(struct audio_port *port) const; |
| 218 | |
| 219 | void loadSamplingRates(char *name); |
| 220 | void loadFormats(char *name); |
| 221 | void loadOutChannels(char *name); |
| 222 | void loadInChannels(char *name); |
| 223 | |
| 224 | audio_gain_mode_t loadGainMode(char *name); |
| 225 | void loadGain(cnode *root); |
| 226 | void loadGains(cnode *root); |
| 227 | |
| 228 | void dump(int fd, int spaces) const; |
| 229 | |
| 230 | String8 mName; |
| 231 | audio_port_type_t mType; |
| 232 | audio_port_role_t mRole; |
| 233 | // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats |
| 234 | // indicates the supported parameters should be read from the output stream |
| 235 | // after it is opened for the first time |
| 236 | Vector <uint32_t> mSamplingRates; // supported sampling rates |
| 237 | Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks |
| 238 | Vector <audio_format_t> mFormats; // supported audio formats |
| 239 | Vector < sp<AudioGain> > mGains; // gain controllers |
| 240 | HwModule *mModule; // audio HW module exposing this I/O stream |
| 241 | }; |
| 242 | |
| 243 | class AudioPatch: public RefBase |
| 244 | { |
| 245 | public: |
| 246 | AudioPatch(audio_patch_handle_t handle, |
| 247 | const struct audio_patch *patch, uid_t uid) : |
| 248 | mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {} |
| 249 | |
| 250 | audio_patch_handle_t mHandle; |
| 251 | struct audio_patch mPatch; |
| 252 | uid_t mUid; |
| 253 | audio_patch_handle_t mAfPatchHandle; |
| 254 | }; |
| 255 | |
| 256 | class DeviceDescriptor: public AudioPort |
| 257 | { |
| 258 | public: |
| 259 | DeviceDescriptor(const String8& name, audio_devices_t type, String8 address, |
| 260 | audio_channel_mask_t channelMask) : |
| 261 | AudioPort(name, AUDIO_PORT_TYPE_DEVICE, |
| 262 | audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : |
| 263 | AUDIO_PORT_ROLE_SOURCE, |
| 264 | NULL), |
| 265 | mDeviceType(type), mAddress(address), |
| 266 | mChannelMask(channelMask), mId(0) {} |
| 267 | |
| 268 | DeviceDescriptor(String8 name, audio_devices_t type) : |
| 269 | AudioPort(name, AUDIO_PORT_TYPE_DEVICE, |
| 270 | audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK : |
| 271 | AUDIO_PORT_ROLE_SOURCE, |
| 272 | NULL), |
| 273 | mDeviceType(type), mAddress(""), |
| 274 | mChannelMask(0), mId(0) {} |
| 275 | virtual ~DeviceDescriptor() {} |
| 276 | |
| 277 | bool equals(const sp<DeviceDescriptor>& other) const; |
| 278 | void toAudioPortConfig(struct audio_port_config *dstConfig, |
| 279 | const struct audio_port_config *srcConfig = NULL) const; |
| 280 | |
| 281 | virtual void toAudioPort(struct audio_port *port) const; |
| 282 | |
| 283 | status_t dump(int fd, int spaces, int index) const; |
| 284 | |
| 285 | audio_devices_t mDeviceType; |
| 286 | String8 mAddress; |
| 287 | audio_channel_mask_t mChannelMask; |
| 288 | audio_port_handle_t mId; |
| 289 | }; |
| 290 | |
| 291 | class DeviceVector : public SortedVector< sp<DeviceDescriptor> > |
| 292 | { |
| 293 | public: |
| 294 | DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} |
| 295 | |
| 296 | ssize_t add(const sp<DeviceDescriptor>& item); |
| 297 | ssize_t remove(const sp<DeviceDescriptor>& item); |
| 298 | ssize_t indexOf(const sp<DeviceDescriptor>& item) const; |
| 299 | |
| 300 | audio_devices_t types() const { return mDeviceTypes; } |
| 301 | |
| 302 | void loadDevicesFromType(audio_devices_t types); |
| 303 | void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); |
| 304 | |
| 305 | sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const; |
| 306 | DeviceVector getDevicesFromType(audio_devices_t types) const; |
| 307 | sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const; |
| 308 | sp<DeviceDescriptor> getDeviceFromName(const String8& name) const; |
| 309 | |
| 310 | private: |
| 311 | void refreshTypes(); |
| 312 | audio_devices_t mDeviceTypes; |
| 313 | }; |
| 314 | |
| 315 | // the IOProfile class describes the capabilities of an output or input stream. |
| 316 | // It is currently assumed that all combination of listed parameters are supported. |
| 317 | // It is used by the policy manager to determine if an output or input is suitable for |
| 318 | // a given use case, open/close it accordingly and connect/disconnect audio tracks |
| 319 | // to/from it. |
| 320 | class IOProfile : public AudioPort |
| 321 | { |
| 322 | public: |
| 323 | IOProfile(const String8& name, audio_port_role_t role, HwModule *module); |
| 324 | virtual ~IOProfile(); |
| 325 | |
| 326 | bool isCompatibleProfile(audio_devices_t device, |
| 327 | uint32_t samplingRate, |
| 328 | audio_format_t format, |
| 329 | audio_channel_mask_t channelMask, |
| 330 | audio_output_flags_t flags) const; |
| 331 | |
| 332 | void dump(int fd); |
| 333 | void log(); |
| 334 | |
| 335 | DeviceVector mSupportedDevices; // supported devices |
| 336 | // (devices this output can be routed to) |
| 337 | audio_output_flags_t mFlags; // attribute flags (e.g primary output, |
| 338 | // direct output...). For outputs only. |
| 339 | }; |
| 340 | |
| 341 | class HwModule { |
| 342 | public: |
| 343 | HwModule(const char *name); |
| 344 | ~HwModule(); |
| 345 | |
| 346 | status_t loadOutput(cnode *root); |
| 347 | status_t loadInput(cnode *root); |
| 348 | status_t loadDevice(cnode *root); |
| 349 | |
| 350 | void dump(int fd); |
| 351 | |
| 352 | const char *const mName; // base name of the audio HW module (primary, a2dp ...) |
| 353 | audio_module_handle_t mHandle; |
| 354 | Vector < sp<IOProfile> > mOutputProfiles; // output profiles exposed by this module |
| 355 | Vector < sp<IOProfile> > mInputProfiles; // input profiles exposed by this module |
| 356 | DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf |
| 357 | |
| 358 | }; |
| 359 | |
| 360 | // default volume curve |
| 361 | static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 362 | // default volume curve for media strategy |
| 363 | static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 364 | // volume curve for media strategy on speakers |
| 365 | static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 366 | // volume curve for sonification strategy on speakers |
| 367 | static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 368 | static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT]; |
| 369 | static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 370 | static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT]; |
| 371 | static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 372 | static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 373 | static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; |
| 374 | // default volume curves per stream and device category. See initializeVolumeCurves() |
| 375 | static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; |
| 376 | |
| 377 | // descriptor for audio outputs. Used to maintain current configuration of each opened audio output |
| 378 | // and keep track of the usage of this output by each audio stream type. |
| 379 | class AudioOutputDescriptor |
| 380 | { |
| 381 | public: |
| 382 | AudioOutputDescriptor(const sp<IOProfile>& profile); |
| 383 | |
| 384 | status_t dump(int fd); |
| 385 | |
| 386 | audio_devices_t device() const; |
| 387 | void changeRefCount(audio_stream_type_t stream, int delta); |
| 388 | |
| 389 | bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } |
| 390 | audio_devices_t supportedDevices(); |
| 391 | uint32_t latency(); |
| 392 | bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc); |
| 393 | bool isActive(uint32_t inPastMs = 0) const; |
| 394 | bool isStreamActive(audio_stream_type_t stream, |
| 395 | uint32_t inPastMs = 0, |
| 396 | nsecs_t sysTime = 0) const; |
| 397 | bool isStrategyActive(routing_strategy strategy, |
| 398 | uint32_t inPastMs = 0, |
| 399 | nsecs_t sysTime = 0) const; |
| 400 | |
| 401 | void toAudioPortConfig(struct audio_port_config *dstConfig, |
| 402 | const struct audio_port_config *srcConfig = NULL) const; |
| 403 | void toAudioPort(struct audio_port *port) const; |
| 404 | |
| 405 | audio_port_handle_t mId; |
| 406 | audio_io_handle_t mIoHandle; // output handle |
| 407 | uint32_t mSamplingRate; // |
| 408 | audio_format_t mFormat; // |
| 409 | audio_channel_mask_t mChannelMask; // output configuration |
| 410 | uint32_t mLatency; // |
| 411 | audio_output_flags_t mFlags; // |
| 412 | audio_devices_t mDevice; // current device this output is routed to |
| 413 | audio_patch_handle_t mPatchHandle; |
| 414 | uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output |
| 415 | nsecs_t mStopTime[AUDIO_STREAM_CNT]; |
| 416 | AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output |
| 417 | AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output |
| 418 | float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume |
| 419 | int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter |
| 420 | const sp<IOProfile> mProfile; // I/O profile this output derives from |
| 421 | bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible |
| 422 | // device selection. See checkDeviceMuteStrategies() |
| 423 | uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) |
| 424 | }; |
| 425 | |
| 426 | // descriptor for audio inputs. Used to maintain current configuration of each opened audio input |
| 427 | // and keep track of the usage of this input. |
| 428 | class AudioInputDescriptor |
| 429 | { |
| 430 | public: |
| 431 | AudioInputDescriptor(const sp<IOProfile>& profile); |
| 432 | |
| 433 | status_t dump(int fd); |
| 434 | |
| 435 | audio_port_handle_t mId; |
| 436 | audio_io_handle_t mIoHandle; // input handle |
| 437 | uint32_t mSamplingRate; // |
| 438 | audio_format_t mFormat; // input configuration |
| 439 | audio_channel_mask_t mChannelMask; // |
| 440 | audio_devices_t mDevice; // current device this input is routed to |
| 441 | audio_patch_handle_t mPatchHandle; |
| 442 | uint32_t mRefCount; // number of AudioRecord clients using this output |
| 443 | audio_source_t mInputSource; // input source selected by application (mediarecorder.h) |
| 444 | const sp<IOProfile> mProfile; // I/O profile this output derives from |
| 445 | |
| 446 | void toAudioPortConfig(struct audio_port_config *dstConfig, |
| 447 | const struct audio_port_config *srcConfig = NULL) const; |
| 448 | void toAudioPort(struct audio_port *port) const; |
| 449 | }; |
| 450 | |
| 451 | // stream descriptor used for volume control |
| 452 | class StreamDescriptor |
| 453 | { |
| 454 | public: |
| 455 | StreamDescriptor(); |
| 456 | |
| 457 | int getVolumeIndex(audio_devices_t device); |
| 458 | void dump(int fd); |
| 459 | |
| 460 | int mIndexMin; // min volume index |
| 461 | int mIndexMax; // max volume index |
| 462 | KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device |
| 463 | bool mCanBeMuted; // true is the stream can be muted |
| 464 | |
| 465 | const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; |
| 466 | }; |
| 467 | |
| 468 | // stream descriptor used for volume control |
| 469 | class EffectDescriptor |
| 470 | { |
| 471 | public: |
| 472 | |
| 473 | status_t dump(int fd); |
| 474 | |
| 475 | int mIo; // io the effect is attached to |
| 476 | routing_strategy mStrategy; // routing strategy the effect is associated to |
| 477 | int mSession; // audio session the effect is on |
| 478 | effect_descriptor_t mDesc; // effect descriptor |
| 479 | bool mEnabled; // enabled state: CPU load being used or not |
| 480 | }; |
| 481 | |
| 482 | void addOutput(audio_io_handle_t output, AudioOutputDescriptor *outputDesc); |
| 483 | void addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc); |
| 484 | |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 485 | // return the strategy corresponding to a given stream type |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 486 | static routing_strategy getStrategy(audio_stream_type_t stream); |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 487 | |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 488 | // return appropriate device for streams handled by the specified strategy according to current |
| 489 | // phone state, connected devices... |
| 490 | // if fromCache is true, the device is returned from mDeviceForStrategy[], |
| 491 | // otherwise it is determine by current state |
| 492 | // (device connected,phone state, force use, a2dp output...) |
| 493 | // This allows to: |
| 494 | // 1 speed up process when the state is stable (when starting or stopping an output) |
| 495 | // 2 access to either current device selection (fromCache == true) or |
| 496 | // "future" device selection (fromCache == false) when called from a context |
| 497 | // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND |
| 498 | // before updateDevicesAndOutputs() is called. |
| 499 | virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 500 | bool fromCache); |
| 501 | |
| 502 | // change the route of the specified output. Returns the number of ms we have slept to |
| 503 | // allow new routing to take effect in certain cases. |
| 504 | uint32_t setOutputDevice(audio_io_handle_t output, |
| 505 | audio_devices_t device, |
| 506 | bool force = false, |
| 507 | int delayMs = 0, |
| 508 | audio_patch_handle_t *patchHandle = NULL); |
| 509 | status_t resetOutputDevice(audio_io_handle_t output, |
| 510 | int delayMs = 0, |
| 511 | audio_patch_handle_t *patchHandle = NULL); |
| 512 | status_t setInputDevice(audio_io_handle_t input, |
| 513 | audio_devices_t device, |
| 514 | bool force = false, |
| 515 | audio_patch_handle_t *patchHandle = NULL); |
| 516 | status_t resetInputDevice(audio_io_handle_t input, |
| 517 | audio_patch_handle_t *patchHandle = NULL); |
| 518 | |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 519 | // select input device corresponding to requested audio source |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 520 | virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource); |
| 521 | |
| 522 | // return io handle of active input or 0 if no input is active |
| 523 | // Only considers inputs from physical devices (e.g. main mic, headset mic) when |
| 524 | // ignoreVirtualInputs is true. |
| 525 | audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); |
| 526 | |
| 527 | // initialize volume curves for each strategy and device category |
| 528 | void initializeVolumeCurves(); |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 529 | |
| 530 | // compute the actual volume for a given stream according to the requested index and a particular |
| 531 | // device |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 532 | virtual float computeVolume(audio_stream_type_t stream, int index, |
| 533 | audio_io_handle_t output, audio_devices_t device); |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 534 | |
| 535 | // check that volume change is permitted, compute and send new volume to audio hardware |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 536 | status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, |
| 537 | audio_devices_t device, int delayMs = 0, bool force = false); |
| 538 | |
| 539 | // apply all stream volumes to the specified output and device |
| 540 | void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); |
| 541 | |
| 542 | // Mute or unmute all streams handled by the specified strategy on the specified output |
| 543 | void setStrategyMute(routing_strategy strategy, |
| 544 | bool on, |
| 545 | audio_io_handle_t output, |
| 546 | int delayMs = 0, |
| 547 | audio_devices_t device = (audio_devices_t)0); |
| 548 | |
| 549 | // Mute or unmute the stream on the specified output |
| 550 | void setStreamMute(audio_stream_type_t stream, |
| 551 | bool on, |
| 552 | audio_io_handle_t output, |
| 553 | int delayMs = 0, |
| 554 | audio_devices_t device = (audio_devices_t)0); |
| 555 | |
| 556 | // handle special cases for sonification strategy while in call: mute streams or replace by |
| 557 | // a special tone in the device used for communication |
| 558 | void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange); |
| 559 | |
| 560 | // true if device is in a telephony or VoIP call |
| 561 | virtual bool isInCall(); |
| 562 | |
| 563 | // true if given state represents a device in a telephony or VoIP call |
| 564 | virtual bool isStateInCall(int state); |
| 565 | |
| 566 | // when a device is connected, checks if an open output can be routed |
| 567 | // to this device. If none is open, tries to open one of the available outputs. |
| 568 | // Returns an output suitable to this device or 0. |
| 569 | // when a device is disconnected, checks if an output is not used any more and |
| 570 | // returns its handle if any. |
| 571 | // transfers the audio tracks and effects from one output thread to another accordingly. |
| 572 | status_t checkOutputsForDevice(audio_devices_t device, |
| 573 | audio_policy_dev_state_t state, |
| 574 | SortedVector<audio_io_handle_t>& outputs, |
| 575 | const String8 address); |
| 576 | |
| 577 | status_t checkInputsForDevice(audio_devices_t device, |
| 578 | audio_policy_dev_state_t state, |
| 579 | SortedVector<audio_io_handle_t>& inputs, |
| 580 | const String8 address); |
| 581 | |
| 582 | // close an output and its companion duplicating output. |
| 583 | void closeOutput(audio_io_handle_t output); |
| 584 | |
| 585 | // checks and if necessary changes outputs used for all strategies. |
| 586 | // must be called every time a condition that affects the output choice for a given strategy |
| 587 | // changes: connected device, phone state, force use... |
| 588 | // Must be called before updateDevicesAndOutputs() |
| 589 | void checkOutputForStrategy(routing_strategy strategy); |
| 590 | |
| 591 | // Same as checkOutputForStrategy() but for a all strategies in order of priority |
| 592 | void checkOutputForAllStrategies(); |
| 593 | |
| 594 | // manages A2DP output suspend/restore according to phone state and BT SCO usage |
| 595 | void checkA2dpSuspend(); |
| 596 | |
| 597 | // returns the A2DP output handle if it is open or 0 otherwise |
| 598 | audio_io_handle_t getA2dpOutput(); |
| 599 | |
| 600 | // selects the most appropriate device on output for current state |
| 601 | // must be called every time a condition that affects the device choice for a given output is |
| 602 | // changed: connected device, phone state, force use, output start, output stop.. |
| 603 | // see getDeviceForStrategy() for the use of fromCache parameter |
| 604 | audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); |
| 605 | |
| 606 | // updates cache of device used by all strategies (mDeviceForStrategy[]) |
| 607 | // must be called every time a condition that affects the device choice for a given strategy is |
| 608 | // changed: connected device, phone state, force use... |
| 609 | // cached values are used by getDeviceForStrategy() if parameter fromCache is true. |
| 610 | // Must be called after checkOutputForAllStrategies() |
| 611 | void updateDevicesAndOutputs(); |
| 612 | |
| 613 | // selects the most appropriate device on input for current state |
| 614 | audio_devices_t getNewInputDevice(audio_io_handle_t input); |
| 615 | |
| 616 | virtual uint32_t getMaxEffectsCpuLoad(); |
| 617 | virtual uint32_t getMaxEffectsMemory(); |
| 618 | #ifdef AUDIO_POLICY_TEST |
| 619 | virtual bool threadLoop(); |
| 620 | void exit(); |
| 621 | int testOutputIndex(audio_io_handle_t output); |
| 622 | #endif //AUDIO_POLICY_TEST |
| 623 | |
| 624 | status_t setEffectEnabled(EffectDescriptor *pDesc, bool enabled); |
Ravi Kumar Alamanda | 88d28cb | 2013-10-15 16:59:57 -0700 | [diff] [blame] | 625 | |
| 626 | // returns the category the device belongs to with regard to volume curve management |
| 627 | static device_category getDeviceCategory(audio_devices_t device); |
| 628 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 629 | // extract one device relevant for volume control from multiple device selection |
| 630 | static audio_devices_t getDeviceForVolume(audio_devices_t device); |
Tanya Finkel | de496d8 | 2014-03-05 23:59:45 +0200 | [diff] [blame] | 631 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 632 | SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, |
| 633 | DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> openOutputs); |
| 634 | bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, |
| 635 | SortedVector<audio_io_handle_t>& outputs2); |
Tanya Finkel | de496d8 | 2014-03-05 23:59:45 +0200 | [diff] [blame] | 636 | |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 637 | // mute/unmute strategies using an incompatible device combination |
| 638 | // if muting, wait for the audio in pcm buffer to be drained before proceeding |
| 639 | // if unmuting, unmute only after the specified delay |
| 640 | // Returns the number of ms waited |
| 641 | uint32_t checkDeviceMuteStrategies(AudioOutputDescriptor *outputDesc, |
| 642 | audio_devices_t prevDevice, |
| 643 | uint32_t delayMs); |
| 644 | |
| 645 | audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs, |
| 646 | audio_output_flags_t flags); |
| 647 | sp<IOProfile> getInputProfile(audio_devices_t device, |
| 648 | uint32_t samplingRate, |
| 649 | audio_format_t format, |
| 650 | audio_channel_mask_t channelMask); |
| 651 | sp<IOProfile> getProfileForDirectOutput(audio_devices_t device, |
| 652 | uint32_t samplingRate, |
| 653 | audio_format_t format, |
| 654 | audio_channel_mask_t channelMask, |
| 655 | audio_output_flags_t flags); |
| 656 | |
| 657 | audio_io_handle_t selectOutputForEffects(const SortedVector<audio_io_handle_t>& outputs); |
| 658 | |
| 659 | bool isNonOffloadableEffectEnabled(); |
| 660 | |
| 661 | status_t addAudioPatch(audio_patch_handle_t handle, |
| 662 | const sp<AudioPatch>& patch); |
| 663 | status_t removeAudioPatch(audio_patch_handle_t handle); |
| 664 | |
| 665 | AudioOutputDescriptor *getOutputFromId(audio_port_handle_t id) const; |
| 666 | AudioInputDescriptor *getInputFromId(audio_port_handle_t id) const; |
| 667 | HwModule *getModuleForDevice(audio_devices_t device) const; |
| 668 | HwModule *getModuleFromName(const char *name) const; |
| 669 | // |
| 670 | // Audio policy configuration file parsing (audio_policy.conf) |
| 671 | // |
| 672 | static uint32_t stringToEnum(const struct StringToEnum *table, |
| 673 | size_t size, |
| 674 | const char *name); |
| 675 | static const char *enumToString(const struct StringToEnum *table, |
| 676 | size_t size, |
| 677 | uint32_t value); |
| 678 | static bool stringToBool(const char *value); |
| 679 | static audio_output_flags_t parseFlagNames(char *name); |
| 680 | static audio_devices_t parseDeviceNames(char *name); |
| 681 | void loadHwModule(cnode *root); |
| 682 | void loadHwModules(cnode *root); |
| 683 | void loadGlobalConfig(cnode *root, HwModule *module); |
| 684 | status_t loadAudioPolicyConfig(const char *path); |
| 685 | void defaultAudioPolicyConfig(void); |
| 686 | |
| 687 | |
| 688 | uid_t mUidCached; |
| 689 | AudioPolicyClientInterface *mpClientInterface; // audio policy client interface |
| 690 | audio_io_handle_t mPrimaryOutput; // primary output handle |
| 691 | // list of descriptors for outputs currently opened |
| 692 | DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; |
| 693 | // copy of mOutputs before setDeviceConnectionState() opens new outputs |
| 694 | // reset to mOutputs when updateDevicesAndOutputs() is called. |
| 695 | DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs; |
| 696 | DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors |
| 697 | DeviceVector mAvailableOutputDevices; // all available output devices |
| 698 | DeviceVector mAvailableInputDevices; // all available input devices |
| 699 | int mPhoneState; // current phone state |
| 700 | audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration |
| 701 | |
| 702 | StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control |
| 703 | bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected |
| 704 | audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; |
| 705 | float mLastVoiceVolume; // last voice volume value sent to audio HAL |
| 706 | |
| 707 | // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units |
| 708 | static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; |
| 709 | // Maximum memory allocated to audio effects in KB |
| 710 | static const uint32_t MAX_EFFECTS_MEMORY = 512; |
| 711 | uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects |
| 712 | uint32_t mTotalEffectsMemory; // current memory used by effects |
| 713 | KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects |
| 714 | bool mA2dpSuspended; // true if A2DP output is suspended |
| 715 | sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time |
| 716 | bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path |
| 717 | // to boost soft sounds, used to adjust volume curves accordingly |
| 718 | |
| 719 | Vector <HwModule *> mHwModules; |
| 720 | volatile int32_t mNextUniqueId; |
| 721 | volatile int32_t mAudioPortGeneration; |
| 722 | |
| 723 | DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches; |
| 724 | |
| 725 | #ifdef AUDIO_POLICY_TEST |
| 726 | Mutex mLock; |
| 727 | Condition mWaitWorkCV; |
| 728 | |
| 729 | int mCurOutput; |
| 730 | bool mDirectOutput; |
| 731 | audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; |
| 732 | int mTestInput; |
| 733 | uint32_t mTestDevice; |
| 734 | uint32_t mTestSamplingRate; |
| 735 | uint32_t mTestFormat; |
| 736 | uint32_t mTestChannels; |
| 737 | uint32_t mTestLatencyMs; |
| 738 | #endif //AUDIO_POLICY_TEST |
Tanya Finkel | de496d8 | 2014-03-05 23:59:45 +0200 | [diff] [blame] | 739 | |
Pavan Chikkala | 785b693 | 2014-03-24 18:58:11 +0530 | [diff] [blame] | 740 | private: |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 741 | static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, |
| 742 | int indexInUi); |
| 743 | // updates device caching and output for streams that can influence the |
| 744 | // routing of notifications |
| 745 | void handleNotificationRoutingForStream(audio_stream_type_t stream); |
| 746 | static bool isVirtualInputDevice(audio_devices_t device); |
| 747 | uint32_t nextUniqueId(); |
| 748 | uint32_t nextAudioPortGeneration(); |
| 749 | uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; } |
| 750 | // converts device address to string sent to audio HAL via setParameters |
| 751 | static String8 addressToParameter(audio_devices_t device, const String8 address); |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 752 | }; |
Mingming Yin | 0670f16 | 2014-06-12 16:05:49 -0700 | [diff] [blame^] | 753 | |
Ravi Kumar Alamanda | 89a8142 | 2013-10-08 23:47:55 -0700 | [diff] [blame] | 754 | }; |