AudioSystem.h revision b52c152d553556b2d227ffc943489de0c60b4b02
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 28namespace android { 29 30typedef void (*audio_error_callback)(status_t err); 31 32class IAudioFlinger; 33class IAudioPolicyService; 34class String8; 35 36class AudioSystem 37{ 38public: 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 getOutputFrameCount(size_t* frameCount, 103 audio_stream_type_t stream); 104 static status_t getOutputLatency(uint32_t* latency, 105 audio_stream_type_t stream); 106 static status_t getSamplingRate(audio_io_handle_t output, 107 audio_stream_type_t streamType, 108 uint32_t* samplingRate); 109 // returns the number of frames per audio HAL write buffer. Corresponds to 110 // audio_stream->get_buffer_size()/audio_stream_frame_size() 111 static status_t getFrameCount(audio_io_handle_t output, 112 audio_stream_type_t stream, 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 audio session ID and return that new ID. 143 // If unable to contact AudioFlinger, returns AUDIO_SESSION_ALLOCATE instead. 144 // FIXME If AudioFlinger were to ever exhaust the session ID namespace, 145 // this method could fail by returning either AUDIO_SESSION_ALLOCATE 146 // or an unspecified existing session ID. 147 static int newAudioSessionId(); 148 149 static void acquireAudioSessionId(int audioSession, pid_t pid); 150 static void releaseAudioSessionId(int audioSession, pid_t pid); 151 152 // types of io configuration change events received with ioConfigChanged() 153 enum io_config_event { 154 OUTPUT_OPENED, 155 OUTPUT_CLOSED, 156 OUTPUT_CONFIG_CHANGED, 157 INPUT_OPENED, 158 INPUT_CLOSED, 159 INPUT_CONFIG_CHANGED, 160 STREAM_CONFIG_CHANGED, 161 NUM_CONFIG_EVENTS 162 }; 163 164 // audio output descriptor used to cache output configurations in client process to avoid 165 // frequent calls through IAudioFlinger 166 class OutputDescriptor { 167 public: 168 OutputDescriptor() 169 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) 170 {} 171 172 uint32_t samplingRate; 173 audio_format_t format; 174 audio_channel_mask_t channelMask; 175 size_t frameCount; 176 uint32_t latency; 177 }; 178 179 // Events used to synchronize actions between audio sessions. 180 // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until 181 // playback is complete on another audio session. 182 // See definitions in MediaSyncEvent.java 183 enum sync_event_t { 184 SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event 185 SYNC_EVENT_NONE = 0, 186 SYNC_EVENT_PRESENTATION_COMPLETE, 187 188 // 189 // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... 190 // 191 SYNC_EVENT_CNT, 192 }; 193 194 // Timeout for synchronous record start. Prevents from blocking the record thread forever 195 // if the trigger event is not fired. 196 static const uint32_t kSyncRecordStartTimeOutMs = 30000; 197 198 // 199 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 200 // 201 static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, 202 const char *device_address); 203 static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 204 const char *device_address); 205 static status_t setPhoneState(audio_mode_t state); 206 static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); 207 static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 208 209 // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), 210 // or release it with releaseOutput(). 211 static audio_io_handle_t getOutput(audio_stream_type_t stream, 212 uint32_t samplingRate = 0, 213 audio_format_t format = AUDIO_FORMAT_DEFAULT, 214 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 215 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 216 const audio_offload_info_t *offloadInfo = NULL); 217 218 static status_t startOutput(audio_io_handle_t output, 219 audio_stream_type_t stream, 220 int session); 221 static status_t stopOutput(audio_io_handle_t output, 222 audio_stream_type_t stream, 223 int session); 224 static void releaseOutput(audio_io_handle_t output); 225 226 // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), 227 // or release it with releaseInput(). 228 static audio_io_handle_t getInput(audio_source_t inputSource, 229 uint32_t samplingRate, 230 audio_format_t format, 231 audio_channel_mask_t channelMask, 232 int sessionId); 233 234 static status_t startInput(audio_io_handle_t input); 235 static status_t stopInput(audio_io_handle_t input); 236 static void releaseInput(audio_io_handle_t input); 237 static status_t initStreamVolume(audio_stream_type_t stream, 238 int indexMin, 239 int indexMax); 240 static status_t setStreamVolumeIndex(audio_stream_type_t stream, 241 int index, 242 audio_devices_t device); 243 static status_t getStreamVolumeIndex(audio_stream_type_t stream, 244 int *index, 245 audio_devices_t device); 246 247 static uint32_t getStrategyForStream(audio_stream_type_t stream); 248 static audio_devices_t getDevicesForStream(audio_stream_type_t stream); 249 250 static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); 251 static status_t registerEffect(const effect_descriptor_t *desc, 252 audio_io_handle_t io, 253 uint32_t strategy, 254 int session, 255 int id); 256 static status_t unregisterEffect(int id); 257 static status_t setEffectEnabled(int id, bool enabled); 258 259 // clear stream to output mapping cache (gStreamOutputMap) 260 // and output configuration cache (gOutputs) 261 static void clearAudioConfigCache(); 262 263 static const sp<IAudioPolicyService>& get_audio_policy_service(); 264 265 // helpers for android.media.AudioManager.getProperty(), see description there for meaning 266 static uint32_t getPrimaryOutputSamplingRate(); 267 static size_t getPrimaryOutputFrameCount(); 268 269 static status_t setLowRamDevice(bool isLowRamDevice); 270 271 // Check if hw offload is possible for given format, stream type, sample rate, 272 // bit rate, duration, video and streaming or offload property is enabled 273 static bool isOffloadSupported(const audio_offload_info_t& info); 274 275 // check presence of audio flinger service. 276 // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise 277 static status_t checkAudioFlinger(); 278 279 /* List available audio ports and their attributes */ 280 static status_t listAudioPorts(audio_port_role_t role, 281 audio_port_type_t type, 282 unsigned int *num_ports, 283 struct audio_port *ports, 284 unsigned int *generation); 285 286 /* Get attributes for a given audio port */ 287 static status_t getAudioPort(struct audio_port *port); 288 289 /* Create an audio patch between several source and sink ports */ 290 static status_t createAudioPatch(const struct audio_patch *patch, 291 audio_patch_handle_t *handle); 292 293 /* Release an audio patch */ 294 static status_t releaseAudioPatch(audio_patch_handle_t handle); 295 296 /* List existing audio patches */ 297 static status_t listAudioPatches(unsigned int *num_patches, 298 struct audio_patch *patches, 299 unsigned int *generation); 300 /* Set audio port configuration */ 301 static status_t setAudioPortConfig(const struct audio_port_config *config); 302 303 // ---------------------------------------------------------------------------- 304 305 class AudioPortCallback : public RefBase 306 { 307 public: 308 309 AudioPortCallback() {} 310 virtual ~AudioPortCallback() {} 311 312 virtual void onAudioPortListUpdate() = 0; 313 virtual void onAudioPatchListUpdate() = 0; 314 virtual void onServiceDied() = 0; 315 316 }; 317 318 static void setAudioPortCallback(sp<AudioPortCallback> callBack); 319 320private: 321 322 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 323 { 324 public: 325 AudioFlingerClient() { 326 } 327 328 // DeathRecipient 329 virtual void binderDied(const wp<IBinder>& who); 330 331 // IAudioFlingerClient 332 333 // indicate a change in the configuration of an output or input: keeps the cached 334 // values for output/input parameters up-to-date in client process 335 virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 336 }; 337 338 class AudioPolicyServiceClient: public IBinder::DeathRecipient, 339 public BnAudioPolicyServiceClient 340 { 341 public: 342 AudioPolicyServiceClient() { 343 } 344 345 // DeathRecipient 346 virtual void binderDied(const wp<IBinder>& who); 347 348 // IAudioPolicyServiceClient 349 virtual void onAudioPortListUpdate(); 350 virtual void onAudioPatchListUpdate(); 351 }; 352 353 static sp<AudioFlingerClient> gAudioFlingerClient; 354 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 355 friend class AudioFlingerClient; 356 friend class AudioPolicyServiceClient; 357 358 static Mutex gLock; 359 static sp<IAudioFlinger> gAudioFlinger; 360 static audio_error_callback gAudioErrorCallback; 361 362 static size_t gInBuffSize; 363 // previous parameters for recording buffer size queries 364 static uint32_t gPrevInSamplingRate; 365 static audio_format_t gPrevInFormat; 366 static audio_channel_mask_t gPrevInChannelMask; 367 368 static sp<IAudioPolicyService> gAudioPolicyService; 369 370 // list of output descriptors containing cached parameters 371 // (sampling rate, framecount, channel count...) 372 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 373 374 static sp<AudioPortCallback> gAudioPortCallback; 375}; 376 377}; // namespace android 378 379#endif /*ANDROID_AUDIOSYSTEM_H_*/ 380