/* * Copyright (C) 2008 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #ifndef ANDROID_AUDIOSYSTEM_H_ #define ANDROID_AUDIOSYSTEM_H_ #include #include #include #include #include #include #include #include namespace android { typedef void (*audio_error_callback)(status_t err); class IAudioFlinger; class IAudioPolicyService; class String8; class AudioSystem { public: /* These are static methods to control the system-wide AudioFlinger * only privileged processes can have access to them */ // mute/unmute microphone static status_t muteMicrophone(bool state); static status_t isMicrophoneMuted(bool *state); // set/get master volume static status_t setMasterVolume(float value); static status_t getMasterVolume(float* volume); // mute/unmute audio outputs static status_t setMasterMute(bool mute); static status_t getMasterMute(bool* mute); // set/get stream volume on specified output static status_t setStreamVolume(audio_stream_type_t stream, float value, audio_io_handle_t output); static status_t getStreamVolume(audio_stream_type_t stream, float* volume, audio_io_handle_t output); // mute/unmute stream static status_t setStreamMute(audio_stream_type_t stream, bool mute); static status_t getStreamMute(audio_stream_type_t stream, bool* mute); // set audio mode in audio hardware static status_t setMode(audio_mode_t mode); // returns true in *state if tracks are active on the specified stream or have been active // in the past inPastMs milliseconds static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); // returns true in *state if tracks are active for what qualifies as remote playback // on the specified stream or have been active in the past inPastMs milliseconds. Remote // playback isn't mutually exclusive with local playback. static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, uint32_t inPastMs); // returns true in *state if a recorder is currently recording with the specified source static status_t isSourceActive(audio_source_t source, bool *state); // set/get audio hardware parameters. The function accepts a list of parameters // key value pairs in the form: key1=value1;key2=value2;... // Some keys are reserved for standard parameters (See AudioParameter class). // The versions with audio_io_handle_t are intended for internal media framework use only. static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); // The versions without audio_io_handle_t are intended for JNI. static status_t setParameters(const String8& keyValuePairs); static String8 getParameters(const String8& keys); static void setErrorCallback(audio_error_callback cb); // helper function to obtain AudioFlinger service handle static const sp get_audio_flinger(); static float linearToLog(int volume); static int logToLinear(float volume); // Returned samplingRate and frameCount output values are guaranteed // to be non-zero if status == NO_ERROR // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputSamplingRate(uint32_t* samplingRate, audio_stream_type_t stream); // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputFrameCount(size_t* frameCount, audio_stream_type_t stream); // FIXME This API assumes a route, and so should be deprecated. static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream); static status_t getSamplingRate(audio_io_handle_t output, uint32_t* samplingRate); // returns the number of frames per audio HAL write buffer. Corresponds to // audio_stream->get_buffer_size()/audio_stream_out_frame_size() static status_t getFrameCount(audio_io_handle_t output, size_t* frameCount); // returns the audio output latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, uint32_t* latency); // return status NO_ERROR implies *buffSize > 0 // FIXME This API assumes a route, and so should deprecated. static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t* buffSize); static status_t setVoiceVolume(float volume); // return the number of audio frames written by AudioFlinger to audio HAL and // audio dsp to DAC since the specified output has exited standby. // returned status (from utils/Errors.h) can be: // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data // - INVALID_OPERATION: Not supported on current hardware platform // - BAD_VALUE: invalid parameter // NOTE: this feature is not supported on all hardware platforms and it is // necessary to check returned status before using the returned values. static status_t getRenderPosition(audio_io_handle_t output, uint32_t *halFrames, uint32_t *dspFrames); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); // Allocate a new unique ID for use as an audio session ID or I/O handle. // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, // this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE // or an unspecified existing unique ID. static audio_unique_id_t newAudioUniqueId(); static void acquireAudioSessionId(int audioSession, pid_t pid); static void releaseAudioSessionId(int audioSession, pid_t pid); // Get the HW synchronization source used for an audio session. // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs // or no HW sync source is used. static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); // types of io configuration change events received with ioConfigChanged() enum io_config_event { OUTPUT_OPENED, OUTPUT_CLOSED, OUTPUT_CONFIG_CHANGED, INPUT_OPENED, INPUT_CLOSED, INPUT_CONFIG_CHANGED, STREAM_CONFIG_CHANGED, NUM_CONFIG_EVENTS }; // audio output descriptor used to cache output configurations in client process to avoid // frequent calls through IAudioFlinger class OutputDescriptor { public: OutputDescriptor() : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) {} uint32_t samplingRate; audio_format_t format; audio_channel_mask_t channelMask; size_t frameCount; uint32_t latency; }; // Events used to synchronize actions between audio sessions. // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until // playback is complete on another audio session. // See definitions in MediaSyncEvent.java enum sync_event_t { SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event SYNC_EVENT_NONE = 0, SYNC_EVENT_PRESENTATION_COMPLETE, // // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... // SYNC_EVENT_CNT, }; // Timeout for synchronous record start. Prevents from blocking the record thread forever // if the trigger event is not fired. static const uint32_t kSyncRecordStartTimeOutMs = 30000; // // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address, const char *device_name); static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), // or release it with releaseOutput(). static audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, const audio_offload_info_t *offloadInfo = NULL); static status_t getOutputForAttr(const audio_attributes_t *attr, audio_io_handle_t *output, audio_session_t session, audio_stream_type_t *stream, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE, const audio_offload_info_t *offloadInfo = NULL); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); static status_t stopOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); static void releaseOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), // or release it with releaseInput(). static status_t getInputForAttr(const audio_attributes_t *attr, audio_io_handle_t *input, audio_session_t session, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_input_flags_t flags); static status_t startInput(audio_io_handle_t input, audio_session_t session); static status_t stopInput(audio_io_handle_t input, audio_session_t session); static void releaseInput(audio_io_handle_t input, audio_session_t session); static status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax); static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device); static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device); static uint32_t getStrategyForStream(audio_stream_type_t stream); static audio_devices_t getDevicesForStream(audio_stream_type_t stream); static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); static status_t registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, uint32_t strategy, int session, int id); static status_t unregisterEffect(int id); static status_t setEffectEnabled(int id, bool enabled); // clear stream to output mapping cache (gStreamOutputMap) // and output configuration cache (gOutputs) static void clearAudioConfigCache(); static const sp get_audio_policy_service(); // helpers for android.media.AudioManager.getProperty(), see description there for meaning static uint32_t getPrimaryOutputSamplingRate(); static size_t getPrimaryOutputFrameCount(); static status_t setLowRamDevice(bool isLowRamDevice); // Check if hw offload is possible for given format, stream type, sample rate, // bit rate, duration, video and streaming or offload property is enabled static bool isOffloadSupported(const audio_offload_info_t& info); // check presence of audio flinger service. // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise static status_t checkAudioFlinger(); /* List available audio ports and their attributes */ static status_t listAudioPorts(audio_port_role_t role, audio_port_type_t type, unsigned int *num_ports, struct audio_port *ports, unsigned int *generation); /* Get attributes for a given audio port */ static status_t getAudioPort(struct audio_port *port); /* Create an audio patch between several source and sink ports */ static status_t createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle); /* Release an audio patch */ static status_t releaseAudioPatch(audio_patch_handle_t handle); /* List existing audio patches */ static status_t listAudioPatches(unsigned int *num_patches, struct audio_patch *patches, unsigned int *generation); /* Set audio port configuration */ static status_t setAudioPortConfig(const struct audio_port_config *config); static status_t acquireSoundTriggerSession(audio_session_t *session, audio_io_handle_t *ioHandle, audio_devices_t *device); static status_t releaseSoundTriggerSession(audio_session_t session); static audio_mode_t getPhoneState(); static status_t registerPolicyMixes(Vector mixes, bool registration); static status_t startAudioSource(const struct audio_port_config *source, const audio_attributes_t *attributes, audio_io_handle_t *handle); static status_t stopAudioSource(audio_io_handle_t handle); // ---------------------------------------------------------------------------- class AudioPortCallback : public RefBase { public: AudioPortCallback() {} virtual ~AudioPortCallback() {} virtual void onAudioPortListUpdate() = 0; virtual void onAudioPatchListUpdate() = 0; virtual void onServiceDied() = 0; }; static status_t addAudioPortCallback(const sp& callBack); static status_t removeAudioPortCallback(const sp& callBack); private: class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient { public: AudioFlingerClient() { } // DeathRecipient virtual void binderDied(const wp& who); // IAudioFlingerClient // indicate a change in the configuration of an output or input: keeps the cached // values for output/input parameters up-to-date in client process virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); }; class AudioPolicyServiceClient: public IBinder::DeathRecipient, public BnAudioPolicyServiceClient { public: AudioPolicyServiceClient() { } status_t addAudioPortCallback(const sp& callBack); status_t removeAudioPortCallback(const sp& callBack); // DeathRecipient virtual void binderDied(const wp& who); // IAudioPolicyServiceClient virtual void onAudioPortListUpdate(); virtual void onAudioPatchListUpdate(); virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state); private: Mutex mLock; Vector > mAudioPortCallbacks; }; static sp gAudioFlingerClient; static sp gAudioPolicyServiceClient; friend class AudioFlingerClient; friend class AudioPolicyServiceClient; static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat, // gPrevInChannelMask and gInBuffSize static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient static sp gAudioFlinger; static audio_error_callback gAudioErrorCallback; static size_t gInBuffSize; // previous parameters for recording buffer size queries static uint32_t gPrevInSamplingRate; static audio_format_t gPrevInFormat; static audio_channel_mask_t gPrevInChannelMask; static sp gAudioPolicyService; // list of output descriptors containing cached parameters // (sampling rate, framecount, channel count...) static DefaultKeyedVector gOutputs; }; }; // namespace android #endif /*ANDROID_AUDIOSYSTEM_H_*/