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/IAudioFlingerClient.h> 22 #include <media/IAudioPolicyServiceClient.h> 23 #include <system/audio.h> 24 #include <system/audio_policy.h> 25 #include <utils/Errors.h> 26 #include <utils/Mutex.h> 27 28 namespace android { 29 30 typedef void (*audio_error_callback)(status_t err); 31 32 class IAudioFlinger; 33 class IAudioPolicyService; 34 class String8; 35 36 class AudioSystem 37 { 38 public: 39 40 /* These are static methods to control the system-wide AudioFlinger 41 * only privileged processes can have access to them 42 */ 43 44 // mute/unmute microphone 45 static status_t muteMicrophone(bool state); 46 static status_t isMicrophoneMuted(bool *state); 47 48 // set/get master volume 49 static status_t setMasterVolume(float value); 50 static status_t getMasterVolume(float* volume); 51 52 // mute/unmute audio outputs 53 static status_t setMasterMute(bool mute); 54 static status_t getMasterMute(bool* mute); 55 56 // set/get stream volume on specified output 57 static status_t setStreamVolume(audio_stream_type_t stream, float value, 58 audio_io_handle_t output); 59 static status_t getStreamVolume(audio_stream_type_t stream, float* volume, 60 audio_io_handle_t output); 61 62 // mute/unmute stream 63 static status_t setStreamMute(audio_stream_type_t stream, bool mute); 64 static status_t getStreamMute(audio_stream_type_t stream, bool* mute); 65 66 // set audio mode in audio hardware 67 static status_t setMode(audio_mode_t mode); 68 69 // returns true in *state if tracks are active on the specified stream or have been active 70 // in the past inPastMs milliseconds 71 static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); 72 // returns true in *state if tracks are active for what qualifies as remote playback 73 // on the specified stream or have been active in the past inPastMs milliseconds. Remote 74 // playback isn't mutually exclusive with local playback. 75 static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, 76 uint32_t inPastMs); 77 // returns true in *state if a recorder is currently recording with the specified source 78 static status_t isSourceActive(audio_source_t source, bool *state); 79 80 // set/get audio hardware parameters. The function accepts a list of parameters 81 // key value pairs in the form: key1=value1;key2=value2;... 82 // Some keys are reserved for standard parameters (See AudioParameter class). 83 // The versions with audio_io_handle_t are intended for internal media framework use only. 84 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 85 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 86 // The versions without audio_io_handle_t are intended for JNI. 87 static status_t setParameters(const String8& keyValuePairs); 88 static String8 getParameters(const String8& keys); 89 90 static void setErrorCallback(audio_error_callback cb); 91 92 // helper function to obtain AudioFlinger service handle 93 static const sp<IAudioFlinger>& get_audio_flinger(); 94 95 static float linearToLog(int volume); 96 static int logToLinear(float volume); 97 98 // Returned samplingRate and frameCount output values are guaranteed 99 // to be non-zero if status == NO_ERROR 100 static status_t getOutputSamplingRate(uint32_t* samplingRate, 101 audio_stream_type_t stream); 102 static status_t getOutputSamplingRateForAttr(uint32_t* samplingRate, 103 const audio_attributes_t *attr); 104 static status_t getOutputFrameCount(size_t* frameCount, 105 audio_stream_type_t stream); 106 static status_t getOutputLatency(uint32_t* latency, 107 audio_stream_type_t stream); 108 static status_t getSamplingRate(audio_io_handle_t output, 109 uint32_t* samplingRate); 110 // returns the number of frames per audio HAL write buffer. Corresponds to 111 // audio_stream->get_buffer_size()/audio_stream_out_frame_size() 112 static status_t getFrameCount(audio_io_handle_t output, 113 size_t* frameCount); 114 // returns the audio output stream latency in ms. Corresponds to 115 // audio_stream_out->get_latency() 116 static status_t getLatency(audio_io_handle_t output, 117 uint32_t* latency); 118 119 static bool routedToA2dpOutput(audio_stream_type_t streamType); 120 121 // return status NO_ERROR implies *buffSize > 0 122 static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 123 audio_channel_mask_t channelMask, size_t* buffSize); 124 125 static status_t setVoiceVolume(float volume); 126 127 // return the number of audio frames written by AudioFlinger to audio HAL and 128 // audio dsp to DAC since the specified output I/O handle has exited standby. 129 // returned status (from utils/Errors.h) can be: 130 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 131 // - INVALID_OPERATION: Not supported on current hardware platform 132 // - BAD_VALUE: invalid parameter 133 // NOTE: this feature is not supported on all hardware platforms and it is 134 // necessary to check returned status before using the returned values. 135 static status_t getRenderPosition(audio_io_handle_t output, 136 uint32_t *halFrames, 137 uint32_t *dspFrames); 138 139 // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid 140 static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); 141 142 // Allocate a new unique ID for use as an audio session ID or I/O handle. 143 // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. 144 // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, 145 // this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE 146 // or an unspecified existing unique ID. 147 static audio_unique_id_t newAudioUniqueId(); 148 149 static void acquireAudioSessionId(int audioSession, pid_t pid); 150 static void releaseAudioSessionId(int audioSession, pid_t pid); 151 152 // Get the HW synchronization source used for an audio session. 153 // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs 154 // or no HW sync source is used. 155 static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 156 157 // types of io configuration change events received with ioConfigChanged() 158 enum io_config_event { 159 OUTPUT_OPENED, 160 OUTPUT_CLOSED, 161 OUTPUT_CONFIG_CHANGED, 162 INPUT_OPENED, 163 INPUT_CLOSED, 164 INPUT_CONFIG_CHANGED, 165 STREAM_CONFIG_CHANGED, 166 NUM_CONFIG_EVENTS 167 }; 168 169 // audio output descriptor used to cache output configurations in client process to avoid 170 // frequent calls through IAudioFlinger 171 class OutputDescriptor { 172 public: OutputDescriptor()173 OutputDescriptor() 174 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) 175 {} 176 177 uint32_t samplingRate; 178 audio_format_t format; 179 audio_channel_mask_t channelMask; 180 size_t frameCount; 181 uint32_t latency; 182 }; 183 184 // Events used to synchronize actions between audio sessions. 185 // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until 186 // playback is complete on another audio session. 187 // See definitions in MediaSyncEvent.java 188 enum sync_event_t { 189 SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event 190 SYNC_EVENT_NONE = 0, 191 SYNC_EVENT_PRESENTATION_COMPLETE, 192 193 // 194 // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... 195 // 196 SYNC_EVENT_CNT, 197 }; 198 199 // Timeout for synchronous record start. Prevents from blocking the record thread forever 200 // if the trigger event is not fired. 201 static const uint32_t kSyncRecordStartTimeOutMs = 30000; 202 203 // 204 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 205 // 206 static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, 207 const char *device_address); 208 static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 209 const char *device_address); 210 static status_t setPhoneState(audio_mode_t state); 211 static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); 212 static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 213 214 // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), 215 // or release it with releaseOutput(). 216 static audio_io_handle_t getOutput(audio_stream_type_t stream, 217 uint32_t samplingRate = 0, 218 audio_format_t format = AUDIO_FORMAT_DEFAULT, 219 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 220 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 221 const audio_offload_info_t *offloadInfo = NULL); 222 static audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr, 223 uint32_t samplingRate = 0, 224 audio_format_t format = AUDIO_FORMAT_DEFAULT, 225 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 226 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 227 const audio_offload_info_t *offloadInfo = NULL); 228 static status_t startOutput(audio_io_handle_t output, 229 audio_stream_type_t stream, 230 int session); 231 static status_t stopOutput(audio_io_handle_t output, 232 audio_stream_type_t stream, 233 int session); 234 static void releaseOutput(audio_io_handle_t output); 235 236 // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), 237 // or release it with releaseInput(). 238 static audio_io_handle_t getInput(audio_source_t inputSource, 239 uint32_t samplingRate, 240 audio_format_t format, 241 audio_channel_mask_t channelMask, 242 int sessionId, 243 audio_input_flags_t); 244 245 static status_t startInput(audio_io_handle_t input, 246 audio_session_t session); 247 static status_t stopInput(audio_io_handle_t input, 248 audio_session_t session); 249 static void releaseInput(audio_io_handle_t input, 250 audio_session_t session); 251 static status_t initStreamVolume(audio_stream_type_t stream, 252 int indexMin, 253 int indexMax); 254 static status_t setStreamVolumeIndex(audio_stream_type_t stream, 255 int index, 256 audio_devices_t device); 257 static status_t getStreamVolumeIndex(audio_stream_type_t stream, 258 int *index, 259 audio_devices_t device); 260 261 static uint32_t getStrategyForStream(audio_stream_type_t stream); 262 static audio_devices_t getDevicesForStream(audio_stream_type_t stream); 263 264 static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); 265 static status_t registerEffect(const effect_descriptor_t *desc, 266 audio_io_handle_t io, 267 uint32_t strategy, 268 int session, 269 int id); 270 static status_t unregisterEffect(int id); 271 static status_t setEffectEnabled(int id, bool enabled); 272 273 // clear stream to output mapping cache (gStreamOutputMap) 274 // and output configuration cache (gOutputs) 275 static void clearAudioConfigCache(); 276 277 static const sp<IAudioPolicyService>& get_audio_policy_service(); 278 279 // helpers for android.media.AudioManager.getProperty(), see description there for meaning 280 static uint32_t getPrimaryOutputSamplingRate(); 281 static size_t getPrimaryOutputFrameCount(); 282 283 static status_t setLowRamDevice(bool isLowRamDevice); 284 285 // Check if hw offload is possible for given format, stream type, sample rate, 286 // bit rate, duration, video and streaming or offload property is enabled 287 static bool isOffloadSupported(const audio_offload_info_t& info); 288 289 // check presence of audio flinger service. 290 // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise 291 static status_t checkAudioFlinger(); 292 293 /* List available audio ports and their attributes */ 294 static status_t listAudioPorts(audio_port_role_t role, 295 audio_port_type_t type, 296 unsigned int *num_ports, 297 struct audio_port *ports, 298 unsigned int *generation); 299 300 /* Get attributes for a given audio port */ 301 static status_t getAudioPort(struct audio_port *port); 302 303 /* Create an audio patch between several source and sink ports */ 304 static status_t createAudioPatch(const struct audio_patch *patch, 305 audio_patch_handle_t *handle); 306 307 /* Release an audio patch */ 308 static status_t releaseAudioPatch(audio_patch_handle_t handle); 309 310 /* List existing audio patches */ 311 static status_t listAudioPatches(unsigned int *num_patches, 312 struct audio_patch *patches, 313 unsigned int *generation); 314 /* Set audio port configuration */ 315 static status_t setAudioPortConfig(const struct audio_port_config *config); 316 317 318 static status_t acquireSoundTriggerSession(audio_session_t *session, 319 audio_io_handle_t *ioHandle, 320 audio_devices_t *device); 321 static status_t releaseSoundTriggerSession(audio_session_t session); 322 323 static audio_mode_t getPhoneState(); 324 325 // ---------------------------------------------------------------------------- 326 327 class AudioPortCallback : public RefBase 328 { 329 public: 330 AudioPortCallback()331 AudioPortCallback() {} ~AudioPortCallback()332 virtual ~AudioPortCallback() {} 333 334 virtual void onAudioPortListUpdate() = 0; 335 virtual void onAudioPatchListUpdate() = 0; 336 virtual void onServiceDied() = 0; 337 338 }; 339 340 static void setAudioPortCallback(sp<AudioPortCallback> callBack); 341 342 private: 343 344 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 345 { 346 public: AudioFlingerClient()347 AudioFlingerClient() { 348 } 349 350 // DeathRecipient 351 virtual void binderDied(const wp<IBinder>& who); 352 353 // IAudioFlingerClient 354 355 // indicate a change in the configuration of an output or input: keeps the cached 356 // values for output/input parameters up-to-date in client process 357 virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 358 }; 359 360 class AudioPolicyServiceClient: public IBinder::DeathRecipient, 361 public BnAudioPolicyServiceClient 362 { 363 public: AudioPolicyServiceClient()364 AudioPolicyServiceClient() { 365 } 366 367 // DeathRecipient 368 virtual void binderDied(const wp<IBinder>& who); 369 370 // IAudioPolicyServiceClient 371 virtual void onAudioPortListUpdate(); 372 virtual void onAudioPatchListUpdate(); 373 }; 374 375 static sp<AudioFlingerClient> gAudioFlingerClient; 376 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 377 friend class AudioFlingerClient; 378 friend class AudioPolicyServiceClient; 379 380 static Mutex gLock; 381 static sp<IAudioFlinger> gAudioFlinger; 382 static audio_error_callback gAudioErrorCallback; 383 384 static size_t gInBuffSize; 385 // previous parameters for recording buffer size queries 386 static uint32_t gPrevInSamplingRate; 387 static audio_format_t gPrevInFormat; 388 static audio_channel_mask_t gPrevInChannelMask; 389 390 static sp<IAudioPolicyService> gAudioPolicyService; 391 392 // list of output descriptors containing cached parameters 393 // (sampling rate, framecount, channel count...) 394 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 395 396 static sp<AudioPortCallback> gAudioPortCallback; 397 }; 398 399 }; // namespace android 400 401 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 402