1 /* 2 * Copyright (C) 2008 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 #ifndef ANDROID_AUDIOSYSTEM_H_ 18 #define ANDROID_AUDIOSYSTEM_H_ 19 20 #include <hardware/audio_effect.h> 21 #include <media/AudioPolicy.h> 22 #include <media/AudioIoDescriptor.h> 23 #include <media/IAudioFlingerClient.h> 24 #include <media/IAudioPolicyServiceClient.h> 25 #include <system/audio.h> 26 #include <system/audio_policy.h> 27 #include <utils/Errors.h> 28 #include <utils/Mutex.h> 29 30 namespace android { 31 32 typedef void (*audio_error_callback)(status_t err); 33 typedef void (*dynamic_policy_callback)(int event, String8 regId, int val); 34 typedef void (*record_config_callback)(int event, audio_session_t session, int source, 35 const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig, 36 audio_patch_handle_t patchHandle); 37 38 class IAudioFlinger; 39 class IAudioPolicyService; 40 class String8; 41 42 class AudioSystem 43 { 44 public: 45 46 // FIXME Declare in binder opcode order, similarly to IAudioFlinger.h and IAudioFlinger.cpp 47 48 /* These are static methods to control the system-wide AudioFlinger 49 * only privileged processes can have access to them 50 */ 51 52 // mute/unmute microphone 53 static status_t muteMicrophone(bool state); 54 static status_t isMicrophoneMuted(bool *state); 55 56 // set/get master volume 57 static status_t setMasterVolume(float value); 58 static status_t getMasterVolume(float* volume); 59 60 // mute/unmute audio outputs 61 static status_t setMasterMute(bool mute); 62 static status_t getMasterMute(bool* mute); 63 64 // set/get stream volume on specified output 65 static status_t setStreamVolume(audio_stream_type_t stream, float value, 66 audio_io_handle_t output); 67 static status_t getStreamVolume(audio_stream_type_t stream, float* volume, 68 audio_io_handle_t output); 69 70 // mute/unmute stream 71 static status_t setStreamMute(audio_stream_type_t stream, bool mute); 72 static status_t getStreamMute(audio_stream_type_t stream, bool* mute); 73 74 // set audio mode in audio hardware 75 static status_t setMode(audio_mode_t mode); 76 77 // returns true in *state if tracks are active on the specified stream or have been active 78 // in the past inPastMs milliseconds 79 static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); 80 // returns true in *state if tracks are active for what qualifies as remote playback 81 // on the specified stream or have been active in the past inPastMs milliseconds. Remote 82 // playback isn't mutually exclusive with local playback. 83 static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, 84 uint32_t inPastMs); 85 // returns true in *state if a recorder is currently recording with the specified source 86 static status_t isSourceActive(audio_source_t source, bool *state); 87 88 // set/get audio hardware parameters. The function accepts a list of parameters 89 // key value pairs in the form: key1=value1;key2=value2;... 90 // Some keys are reserved for standard parameters (See AudioParameter class). 91 // The versions with audio_io_handle_t are intended for internal media framework use only. 92 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 93 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 94 // The versions without audio_io_handle_t are intended for JNI. 95 static status_t setParameters(const String8& keyValuePairs); 96 static String8 getParameters(const String8& keys); 97 98 static void setErrorCallback(audio_error_callback cb); 99 static void setDynPolicyCallback(dynamic_policy_callback cb); 100 static void setRecordConfigCallback(record_config_callback); 101 102 // helper function to obtain AudioFlinger service handle 103 static const sp<IAudioFlinger> get_audio_flinger(); 104 105 static float linearToLog(int volume); 106 static int logToLinear(float volume); 107 108 // Returned samplingRate and frameCount output values are guaranteed 109 // to be non-zero if status == NO_ERROR 110 // FIXME This API assumes a route, and so should be deprecated. 111 static status_t getOutputSamplingRate(uint32_t* samplingRate, 112 audio_stream_type_t stream); 113 // FIXME This API assumes a route, and so should be deprecated. 114 static status_t getOutputFrameCount(size_t* frameCount, 115 audio_stream_type_t stream); 116 // FIXME This API assumes a route, and so should be deprecated. 117 static status_t getOutputLatency(uint32_t* latency, 118 audio_stream_type_t stream); 119 // returns the audio HAL sample rate 120 static status_t getSamplingRate(audio_io_handle_t ioHandle, 121 uint32_t* samplingRate); 122 // For output threads with a fast mixer, returns the number of frames per normal mixer buffer. 123 // For output threads without a fast mixer, or for input, this is same as getFrameCountHAL(). 124 static status_t getFrameCount(audio_io_handle_t ioHandle, 125 size_t* frameCount); 126 // returns the audio output latency in ms. Corresponds to 127 // audio_stream_out->get_latency() 128 static status_t getLatency(audio_io_handle_t output, 129 uint32_t* latency); 130 131 // return status NO_ERROR implies *buffSize > 0 132 // FIXME This API assumes a route, and so should deprecated. 133 static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 134 audio_channel_mask_t channelMask, size_t* buffSize); 135 136 static status_t setVoiceVolume(float volume); 137 138 // return the number of audio frames written by AudioFlinger to audio HAL and 139 // audio dsp to DAC since the specified output has exited standby. 140 // returned status (from utils/Errors.h) can be: 141 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 142 // - INVALID_OPERATION: Not supported on current hardware platform 143 // - BAD_VALUE: invalid parameter 144 // NOTE: this feature is not supported on all hardware platforms and it is 145 // necessary to check returned status before using the returned values. 146 static status_t getRenderPosition(audio_io_handle_t output, 147 uint32_t *halFrames, 148 uint32_t *dspFrames); 149 150 // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid 151 static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); 152 153 // Allocate a new unique ID for use as an audio session ID or I/O handle. 154 // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. 155 // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, 156 // this method could fail by returning either a reserved ID like AUDIO_UNIQUE_ID_ALLOCATE 157 // or an unspecified existing unique ID. 158 static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 159 160 static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 161 static void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 162 163 // Get the HW synchronization source used for an audio session. 164 // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs 165 // or no HW sync source is used. 166 static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 167 168 // Indicate JAVA services are ready (scheduling, power management ...) 169 static status_t systemReady(); 170 171 // Returns the number of frames per audio HAL buffer. 172 // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input. 173 // See also getFrameCount(). 174 static status_t getFrameCountHAL(audio_io_handle_t ioHandle, 175 size_t* frameCount); 176 177 // Events used to synchronize actions between audio sessions. 178 // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until 179 // playback is complete on another audio session. 180 // See definitions in MediaSyncEvent.java 181 enum sync_event_t { 182 SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event 183 SYNC_EVENT_NONE = 0, 184 SYNC_EVENT_PRESENTATION_COMPLETE, 185 186 // 187 // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... 188 // 189 SYNC_EVENT_CNT, 190 }; 191 192 // Timeout for synchronous record start. Prevents from blocking the record thread forever 193 // if the trigger event is not fired. 194 static const uint32_t kSyncRecordStartTimeOutMs = 30000; 195 196 // 197 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 198 // 199 static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, 200 const char *device_address, const char *device_name); 201 static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 202 const char *device_address); 203 static status_t setPhoneState(audio_mode_t state); 204 static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); 205 static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 206 207 // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), 208 // or release it with releaseOutput(). 209 static audio_io_handle_t getOutput(audio_stream_type_t stream, 210 uint32_t samplingRate = 0, 211 audio_format_t format = AUDIO_FORMAT_DEFAULT, 212 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 213 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 214 const audio_offload_info_t *offloadInfo = NULL); 215 static status_t getOutputForAttr(const audio_attributes_t *attr, 216 audio_io_handle_t *output, 217 audio_session_t session, 218 audio_stream_type_t *stream, 219 uid_t uid, 220 uint32_t samplingRate = 0, 221 audio_format_t format = AUDIO_FORMAT_DEFAULT, 222 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 223 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 224 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, 225 const audio_offload_info_t *offloadInfo = NULL); 226 static status_t startOutput(audio_io_handle_t output, 227 audio_stream_type_t stream, 228 audio_session_t session); 229 static status_t stopOutput(audio_io_handle_t output, 230 audio_stream_type_t stream, 231 audio_session_t session); 232 static void releaseOutput(audio_io_handle_t output, 233 audio_stream_type_t stream, 234 audio_session_t session); 235 236 // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), 237 // or release it with releaseInput(). 238 static status_t getInputForAttr(const audio_attributes_t *attr, 239 audio_io_handle_t *input, 240 audio_session_t session, 241 pid_t pid, 242 uid_t uid, 243 uint32_t samplingRate, 244 audio_format_t format, 245 audio_channel_mask_t channelMask, 246 audio_input_flags_t flags, 247 audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); 248 249 static status_t startInput(audio_io_handle_t input, 250 audio_session_t session); 251 static status_t stopInput(audio_io_handle_t input, 252 audio_session_t session); 253 static void releaseInput(audio_io_handle_t input, 254 audio_session_t session); 255 static status_t initStreamVolume(audio_stream_type_t stream, 256 int indexMin, 257 int indexMax); 258 static status_t setStreamVolumeIndex(audio_stream_type_t stream, 259 int index, 260 audio_devices_t device); 261 static status_t getStreamVolumeIndex(audio_stream_type_t stream, 262 int *index, 263 audio_devices_t device); 264 265 static uint32_t getStrategyForStream(audio_stream_type_t stream); 266 static audio_devices_t getDevicesForStream(audio_stream_type_t stream); 267 268 static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); 269 static status_t registerEffect(const effect_descriptor_t *desc, 270 audio_io_handle_t io, 271 uint32_t strategy, 272 audio_session_t session, 273 int id); 274 static status_t unregisterEffect(int id); 275 static status_t setEffectEnabled(int id, bool enabled); 276 277 // clear stream to output mapping cache (gStreamOutputMap) 278 // and output configuration cache (gOutputs) 279 static void clearAudioConfigCache(); 280 281 static const sp<IAudioPolicyService> get_audio_policy_service(); 282 283 // helpers for android.media.AudioManager.getProperty(), see description there for meaning 284 static uint32_t getPrimaryOutputSamplingRate(); 285 static size_t getPrimaryOutputFrameCount(); 286 287 static status_t setLowRamDevice(bool isLowRamDevice); 288 289 // Check if hw offload is possible for given format, stream type, sample rate, 290 // bit rate, duration, video and streaming or offload property is enabled 291 static bool isOffloadSupported(const audio_offload_info_t& info); 292 293 // check presence of audio flinger service. 294 // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise 295 static status_t checkAudioFlinger(); 296 297 /* List available audio ports and their attributes */ 298 static status_t listAudioPorts(audio_port_role_t role, 299 audio_port_type_t type, 300 unsigned int *num_ports, 301 struct audio_port *ports, 302 unsigned int *generation); 303 304 /* Get attributes for a given audio port */ 305 static status_t getAudioPort(struct audio_port *port); 306 307 /* Create an audio patch between several source and sink ports */ 308 static status_t createAudioPatch(const struct audio_patch *patch, 309 audio_patch_handle_t *handle); 310 311 /* Release an audio patch */ 312 static status_t releaseAudioPatch(audio_patch_handle_t handle); 313 314 /* List existing audio patches */ 315 static status_t listAudioPatches(unsigned int *num_patches, 316 struct audio_patch *patches, 317 unsigned int *generation); 318 /* Set audio port configuration */ 319 static status_t setAudioPortConfig(const struct audio_port_config *config); 320 321 322 static status_t acquireSoundTriggerSession(audio_session_t *session, 323 audio_io_handle_t *ioHandle, 324 audio_devices_t *device); 325 static status_t releaseSoundTriggerSession(audio_session_t session); 326 327 static audio_mode_t getPhoneState(); 328 329 static status_t registerPolicyMixes(Vector<AudioMix> mixes, bool registration); 330 331 static status_t startAudioSource(const struct audio_port_config *source, 332 const audio_attributes_t *attributes, 333 audio_io_handle_t *handle); 334 static status_t stopAudioSource(audio_io_handle_t handle); 335 336 static status_t setMasterMono(bool mono); 337 static status_t getMasterMono(bool *mono); 338 339 // ---------------------------------------------------------------------------- 340 341 class AudioPortCallback : public RefBase 342 { 343 public: 344 345 AudioPortCallback() {} 346 virtual ~AudioPortCallback() {} 347 348 virtual void onAudioPortListUpdate() = 0; 349 virtual void onAudioPatchListUpdate() = 0; 350 virtual void onServiceDied() = 0; 351 352 }; 353 354 static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback); 355 static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback); 356 357 class AudioDeviceCallback : public RefBase 358 { 359 public: 360 361 AudioDeviceCallback() {} 362 virtual ~AudioDeviceCallback() {} 363 364 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 365 audio_port_handle_t deviceId) = 0; 366 }; 367 368 static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, 369 audio_io_handle_t audioIo); 370 static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, 371 audio_io_handle_t audioIo); 372 373 static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); 374 375 private: 376 377 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 378 { 379 public: 380 AudioFlingerClient() : 381 mInBuffSize(0), mInSamplingRate(0), 382 mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) { 383 } 384 385 void clearIoCache(); 386 status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 387 audio_channel_mask_t channelMask, size_t* buffSize); 388 sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); 389 390 // DeathRecipient 391 virtual void binderDied(const wp<IBinder>& who); 392 393 // IAudioFlingerClient 394 395 // indicate a change in the configuration of an output or input: keeps the cached 396 // values for output/input parameters up-to-date in client process 397 virtual void ioConfigChanged(audio_io_config_event event, 398 const sp<AudioIoDescriptor>& ioDesc); 399 400 401 status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, 402 audio_io_handle_t audioIo); 403 status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, 404 audio_io_handle_t audioIo); 405 406 audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); 407 408 private: 409 Mutex mLock; 410 DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> > mIoDescriptors; 411 DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > > 412 mAudioDeviceCallbacks; 413 // cached values for recording getInputBufferSize() queries 414 size_t mInBuffSize; // zero indicates cache is invalid 415 uint32_t mInSamplingRate; 416 audio_format_t mInFormat; 417 audio_channel_mask_t mInChannelMask; 418 sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle); 419 }; 420 421 class AudioPolicyServiceClient: public IBinder::DeathRecipient, 422 public BnAudioPolicyServiceClient 423 { 424 public: 425 AudioPolicyServiceClient() { 426 } 427 428 int addAudioPortCallback(const sp<AudioPortCallback>& callback); 429 int removeAudioPortCallback(const sp<AudioPortCallback>& callback); 430 431 // DeathRecipient 432 virtual void binderDied(const wp<IBinder>& who); 433 434 // IAudioPolicyServiceClient 435 virtual void onAudioPortListUpdate(); 436 virtual void onAudioPatchListUpdate(); 437 virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state); 438 virtual void onRecordingConfigurationUpdate(int event, audio_session_t session, 439 audio_source_t source, const audio_config_base_t *clientConfig, 440 const audio_config_base_t *deviceConfig, audio_patch_handle_t patchHandle); 441 442 private: 443 Mutex mLock; 444 Vector <sp <AudioPortCallback> > mAudioPortCallbacks; 445 }; 446 447 static const sp<AudioFlingerClient> getAudioFlingerClient(); 448 static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); 449 450 static sp<AudioFlingerClient> gAudioFlingerClient; 451 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 452 friend class AudioFlingerClient; 453 friend class AudioPolicyServiceClient; 454 455 static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, 456 static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient 457 static sp<IAudioFlinger> gAudioFlinger; 458 static audio_error_callback gAudioErrorCallback; 459 static dynamic_policy_callback gDynPolicyCallback; 460 static record_config_callback gRecordConfigCallback; 461 462 static size_t gInBuffSize; 463 // previous parameters for recording buffer size queries 464 static uint32_t gPrevInSamplingRate; 465 static audio_format_t gPrevInFormat; 466 static audio_channel_mask_t gPrevInChannelMask; 467 468 static sp<IAudioPolicyService> gAudioPolicyService; 469 }; 470 471 }; // namespace android 472 473 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 474
