| /* |
| * 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 <hardware/audio_effect.h> |
| #include <media/AudioPolicy.h> |
| #include <media/AudioIoDescriptor.h> |
| #include <media/IAudioFlingerClient.h> |
| #include <media/IAudioPolicyServiceClient.h> |
| #include <system/audio.h> |
| #include <system/audio_policy.h> |
| #include <utils/Errors.h> |
| #include <utils/Mutex.h> |
| |
| namespace android { |
| |
| typedef void (*audio_error_callback)(status_t err); |
| typedef void (*dynamic_policy_callback)(int event, String8 regId, int val); |
| typedef void (*record_config_callback)(int event, audio_session_t session, int source, |
| const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig, |
| audio_patch_handle_t patchHandle); |
| |
| class IAudioFlinger; |
| class IAudioPolicyService; |
| class String8; |
| |
| class AudioSystem |
| { |
| public: |
| |
| // FIXME Declare in binder opcode order, similarly to IAudioFlinger.h and IAudioFlinger.cpp |
| |
| /* 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); |
| static void setDynPolicyCallback(dynamic_policy_callback cb); |
| static void setRecordConfigCallback(record_config_callback); |
| |
| // helper function to obtain AudioFlinger service handle |
| static const sp<IAudioFlinger> 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); |
| // returns the audio HAL sample rate |
| static status_t getSamplingRate(audio_io_handle_t ioHandle, |
| uint32_t* samplingRate); |
| // For output threads with a fast mixer, returns the number of frames per normal mixer buffer. |
| // For output threads without a fast mixer, or for input, this is same as getFrameCountHAL(). |
| static status_t getFrameCount(audio_io_handle_t ioHandle, |
| 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 a reserved ID like AUDIO_UNIQUE_ID_ALLOCATE |
| // or an unspecified existing unique ID. |
| static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); |
| |
| static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); |
| static void releaseAudioSessionId(audio_session_t 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); |
| |
| // Indicate JAVA services are ready (scheduling, power management ...) |
| static status_t systemReady(); |
| |
| // Returns the number of frames per audio HAL buffer. |
| // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input. |
| // See also getFrameCount(). |
| static status_t getFrameCountHAL(audio_io_handle_t ioHandle, |
| size_t* frameCount); |
| |
| // 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, |
| uid_t uid, |
| 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, |
| pid_t pid, |
| uid_t uid, |
| uint32_t samplingRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| audio_input_flags_t flags, |
| audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE); |
| |
| 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, |
| audio_session_t 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<IAudioPolicyService> 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<AudioMix> 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); |
| |
| static status_t setMasterMono(bool mono); |
| static status_t getMasterMono(bool *mono); |
| |
| // ---------------------------------------------------------------------------- |
| |
| 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<AudioPortCallback>& callback); |
| static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback); |
| |
| class AudioDeviceCallback : public RefBase |
| { |
| public: |
| |
| AudioDeviceCallback() {} |
| virtual ~AudioDeviceCallback() {} |
| |
| virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, |
| audio_port_handle_t deviceId) = 0; |
| }; |
| |
| static status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, |
| audio_io_handle_t audioIo); |
| static status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, |
| audio_io_handle_t audioIo); |
| |
| static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); |
| |
| private: |
| |
| class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient |
| { |
| public: |
| AudioFlingerClient() : |
| mInBuffSize(0), mInSamplingRate(0), |
| mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) { |
| } |
| |
| void clearIoCache(); |
| status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, |
| audio_channel_mask_t channelMask, size_t* buffSize); |
| sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); |
| |
| // DeathRecipient |
| virtual void binderDied(const wp<IBinder>& 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(audio_io_config_event event, |
| const sp<AudioIoDescriptor>& ioDesc); |
| |
| |
| status_t addAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, |
| audio_io_handle_t audioIo); |
| status_t removeAudioDeviceCallback(const sp<AudioDeviceCallback>& callback, |
| audio_io_handle_t audioIo); |
| |
| audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); |
| |
| private: |
| Mutex mLock; |
| DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> > mIoDescriptors; |
| DefaultKeyedVector<audio_io_handle_t, Vector < sp<AudioDeviceCallback> > > |
| mAudioDeviceCallbacks; |
| // cached values for recording getInputBufferSize() queries |
| size_t mInBuffSize; // zero indicates cache is invalid |
| uint32_t mInSamplingRate; |
| audio_format_t mInFormat; |
| audio_channel_mask_t mInChannelMask; |
| sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle); |
| }; |
| |
| class AudioPolicyServiceClient: public IBinder::DeathRecipient, |
| public BnAudioPolicyServiceClient |
| { |
| public: |
| AudioPolicyServiceClient() { |
| } |
| |
| int addAudioPortCallback(const sp<AudioPortCallback>& callback); |
| int removeAudioPortCallback(const sp<AudioPortCallback>& callback); |
| |
| // DeathRecipient |
| virtual void binderDied(const wp<IBinder>& who); |
| |
| // IAudioPolicyServiceClient |
| virtual void onAudioPortListUpdate(); |
| virtual void onAudioPatchListUpdate(); |
| virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state); |
| virtual void onRecordingConfigurationUpdate(int event, audio_session_t session, |
| audio_source_t source, const audio_config_base_t *clientConfig, |
| const audio_config_base_t *deviceConfig, audio_patch_handle_t patchHandle); |
| |
| private: |
| Mutex mLock; |
| Vector <sp <AudioPortCallback> > mAudioPortCallbacks; |
| }; |
| |
| static const sp<AudioFlingerClient> getAudioFlingerClient(); |
| static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); |
| |
| static sp<AudioFlingerClient> gAudioFlingerClient; |
| static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; |
| friend class AudioFlingerClient; |
| friend class AudioPolicyServiceClient; |
| |
| static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, |
| static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient |
| static sp<IAudioFlinger> gAudioFlinger; |
| static audio_error_callback gAudioErrorCallback; |
| static dynamic_policy_callback gDynPolicyCallback; |
| static record_config_callback gRecordConfigCallback; |
| |
| 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<IAudioPolicyService> gAudioPolicyService; |
| }; |
| |
| }; // namespace android |
| |
| #endif /*ANDROID_AUDIOSYSTEM_H_*/ |
