/* * Copyright (C) 2009 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. */ #include #include #include #include #include #include #include #include #include #include "AudioPolicyInterface.h" namespace android { // ---------------------------------------------------------------------------- // Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB #define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 // Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB #define SONIFICATION_HEADSET_VOLUME_MIN 0.016 // Time in milliseconds during which we consider that music is still active after a music // track was stopped - see computeVolume() #define SONIFICATION_HEADSET_MUSIC_DELAY 5000 // Time in milliseconds after media stopped playing during which we consider that the // sonification should be as unobtrusive as during the time media was playing. #define SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY 5000 // Time in milliseconds during witch some streams are muted while the audio path // is switched #define MUTE_TIME_MS 2000 #define NUM_TEST_OUTPUTS 5 #define NUM_VOL_CURVE_KNEES 2 // Default minimum length allowed for offloading a compressed track // Can be overridden by the audio.offload.min.duration.secs property #define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 #define MAX_MIXER_SAMPLING_RATE 48000 #define MAX_MIXER_CHANNEL_COUNT 8 // ---------------------------------------------------------------------------- // AudioPolicyManager implements audio policy manager behavior common to all platforms. // ---------------------------------------------------------------------------- class AudioPolicyManager: public AudioPolicyInterface #ifdef AUDIO_POLICY_TEST , public Thread #endif //AUDIO_POLICY_TEST { public: AudioPolicyManager(AudioPolicyClientInterface *clientInterface); virtual ~AudioPolicyManager(); // AudioPolicyInterface virtual status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address); virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); virtual void setPhoneState(audio_mode_t state); virtual void setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); virtual void setSystemProperty(const char* property, const char* value); virtual status_t initCheck(); virtual audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo); virtual 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, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo); virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); virtual status_t stopOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); virtual void releaseOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session); virtual 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); // indicates to the audio policy manager that the input starts being used. virtual status_t startInput(audio_io_handle_t input, audio_session_t session); // indicates to the audio policy manager that the input stops being used. virtual status_t stopInput(audio_io_handle_t input, audio_session_t session); virtual void releaseInput(audio_io_handle_t input, audio_session_t session); virtual void closeAllInputs(); virtual void initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax); virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device); virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device); // return the strategy corresponding to a given stream type virtual uint32_t getStrategyForStream(audio_stream_type_t stream); // return the strategy corresponding to the given audio attributes virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr); // return the enabled output devices for the given stream type virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL); virtual status_t registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, uint32_t strategy, int session, int id); virtual status_t unregisterEffect(int id); virtual status_t setEffectEnabled(int id, bool enabled); virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const; // return whether a stream is playing remotely, override to change the definition of // local/remote playback, used for instance by notification manager to not make // media players lose audio focus when not playing locally // For the base implementation, "remotely" means playing during screen mirroring which // uses an output for playback with a non-empty, non "0" address. virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; virtual bool isSourceActive(audio_source_t source) const; virtual status_t dump(int fd); virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo); virtual status_t listAudioPorts(audio_port_role_t role, audio_port_type_t type, unsigned int *num_ports, struct audio_port *ports, unsigned int *generation); virtual status_t getAudioPort(struct audio_port *port); virtual status_t createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid); virtual status_t releaseAudioPatch(audio_patch_handle_t handle, uid_t uid); virtual status_t listAudioPatches(unsigned int *num_patches, struct audio_patch *patches, unsigned int *generation); virtual status_t setAudioPortConfig(const struct audio_port_config *config); virtual void clearAudioPatches(uid_t uid); virtual status_t acquireSoundTriggerSession(audio_session_t *session, audio_io_handle_t *ioHandle, audio_devices_t *device); virtual status_t releaseSoundTriggerSession(audio_session_t session); virtual status_t registerPolicyMixes(Vector mixes); virtual status_t unregisterPolicyMixes(Vector mixes); protected: enum routing_strategy { STRATEGY_MEDIA, STRATEGY_PHONE, STRATEGY_SONIFICATION, STRATEGY_SONIFICATION_RESPECTFUL, STRATEGY_DTMF, STRATEGY_ENFORCED_AUDIBLE, STRATEGY_TRANSMITTED_THROUGH_SPEAKER, STRATEGY_ACCESSIBILITY, STRATEGY_REROUTING, NUM_STRATEGIES }; // 4 points to define the volume attenuation curve, each characterized by the volume // index (from 0 to 100) at which they apply, and the attenuation in dB at that index. // we use 100 steps to avoid rounding errors when computing the volume in volIndexToAmpl() enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3, VOLCNT = 4}; class VolumeCurvePoint { public: int mIndex; float mDBAttenuation; }; // device categories used for volume curve management. enum device_category { DEVICE_CATEGORY_HEADSET, DEVICE_CATEGORY_SPEAKER, DEVICE_CATEGORY_EARPIECE, DEVICE_CATEGORY_EXT_MEDIA, DEVICE_CATEGORY_CNT }; class HwModule; class AudioGain: public RefBase { public: AudioGain(int index, bool useInChannelMask); virtual ~AudioGain() {} void dump(int fd, int spaces, int index) const; void getDefaultConfig(struct audio_gain_config *config); status_t checkConfig(const struct audio_gain_config *config); int mIndex; struct audio_gain mGain; bool mUseInChannelMask; }; class AudioPort: public virtual RefBase { public: AudioPort(const String8& name, audio_port_type_t type, audio_port_role_t role, const sp& module); virtual ~AudioPort() {} virtual void toAudioPort(struct audio_port *port) const; void importAudioPort(const sp port); void clearCapabilities(); void loadSamplingRates(char *name); void loadFormats(char *name); void loadOutChannels(char *name); void loadInChannels(char *name); audio_gain_mode_t loadGainMode(char *name); void loadGain(cnode *root, int index); void loadGains(cnode *root); // searches for an exact match status_t checkExactSamplingRate(uint32_t samplingRate) const; // searches for a compatible match, and returns the best match via updatedSamplingRate status_t checkCompatibleSamplingRate(uint32_t samplingRate, uint32_t *updatedSamplingRate) const; // searches for an exact match status_t checkExactChannelMask(audio_channel_mask_t channelMask) const; // searches for a compatible match, currently implemented for input channel masks only status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const; status_t checkFormat(audio_format_t format) const; status_t checkGain(const struct audio_gain_config *gainConfig, int index) const; uint32_t pickSamplingRate() const; audio_channel_mask_t pickChannelMask() const; audio_format_t pickFormat() const; static const audio_format_t sPcmFormatCompareTable[]; static int compareFormats(audio_format_t format1, audio_format_t format2); void dump(int fd, int spaces) const; String8 mName; audio_port_type_t mType; audio_port_role_t mRole; bool mUseInChannelMask; // by convention, "0' in the first entry in mSamplingRates, mChannelMasks or mFormats // indicates the supported parameters should be read from the output stream // after it is opened for the first time Vector mSamplingRates; // supported sampling rates Vector mChannelMasks; // supported channel masks Vector mFormats; // supported audio formats Vector < sp > mGains; // gain controllers sp mModule; // audio HW module exposing this I/O stream uint32_t mFlags; // attribute flags (e.g primary output, // direct output...). }; class AudioPortConfig: public virtual RefBase { public: AudioPortConfig(); virtual ~AudioPortConfig() {} status_t applyAudioPortConfig(const struct audio_port_config *config, struct audio_port_config *backupConfig = NULL); virtual void toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig = NULL) const = 0; virtual sp getAudioPort() const = 0; uint32_t mSamplingRate; audio_format_t mFormat; audio_channel_mask_t mChannelMask; struct audio_gain_config mGain; }; class AudioPatch: public RefBase { public: AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid) : mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0) {} status_t dump(int fd, int spaces, int index) const; audio_patch_handle_t mHandle; struct audio_patch mPatch; uid_t mUid; audio_patch_handle_t mAfPatchHandle; }; class DeviceDescriptor: public AudioPort, public AudioPortConfig { public: DeviceDescriptor(const String8& name, audio_devices_t type); virtual ~DeviceDescriptor() {} bool equals(const sp& other) const; virtual void toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig = NULL) const; virtual sp getAudioPort() const { return (AudioPort*) this; } virtual void toAudioPort(struct audio_port *port) const; status_t dump(int fd, int spaces, int index) const; audio_devices_t mDeviceType; String8 mAddress; audio_port_handle_t mId; }; class DeviceVector : public SortedVector< sp > { public: DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} ssize_t add(const sp& item); ssize_t remove(const sp& item); ssize_t indexOf(const sp& item) const; audio_devices_t types() const { return mDeviceTypes; } void loadDevicesFromType(audio_devices_t types); void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); sp getDevice(audio_devices_t type, String8 address) const; DeviceVector getDevicesFromType(audio_devices_t types) const; sp getDeviceFromId(audio_port_handle_t id) const; sp getDeviceFromName(const String8& name) const; DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address) const; private: void refreshTypes(); audio_devices_t mDeviceTypes; }; // the IOProfile class describes the capabilities of an output or input stream. // It is currently assumed that all combination of listed parameters are supported. // It is used by the policy manager to determine if an output or input is suitable for // a given use case, open/close it accordingly and connect/disconnect audio tracks // to/from it. class IOProfile : public AudioPort { public: IOProfile(const String8& name, audio_port_role_t role, const sp& module); virtual ~IOProfile(); // This method is used for both output and input. // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate. // For input, flags is interpreted as audio_input_flags_t. // TODO: merge audio_output_flags_t and audio_input_flags_t. bool isCompatibleProfile(audio_devices_t device, String8 address, uint32_t samplingRate, uint32_t *updatedSamplingRate, audio_format_t format, audio_channel_mask_t channelMask, uint32_t flags) const; void dump(int fd); void log(); DeviceVector mSupportedDevices; // supported devices // (devices this output can be routed to) }; class HwModule : public RefBase { public: HwModule(const char *name); ~HwModule(); status_t loadOutput(cnode *root); status_t loadInput(cnode *root); status_t loadDevice(cnode *root); status_t addOutputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address); status_t removeOutputProfile(String8 name); status_t addInputProfile(String8 name, const audio_config_t *config, audio_devices_t device, String8 address); status_t removeInputProfile(String8 name); void dump(int fd); const char *const mName; // base name of the audio HW module (primary, a2dp ...) uint32_t mHalVersion; // audio HAL API version audio_module_handle_t mHandle; Vector < sp > mOutputProfiles; // output profiles exposed by this module Vector < sp > mInputProfiles; // input profiles exposed by this module DeviceVector mDeclaredDevices; // devices declared in audio_policy.conf }; // default volume curve static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManager::VOLCNT]; // default volume curve for media strategy static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT]; // volume curve for non-media audio on ext media outputs (HDMI, Line, etc) static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT]; // volume curve for media strategy on speakers static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT]; // volume curve for sonification strategy on speakers static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT]; static const VolumeCurvePoint sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT]; // default volume curves per stream and device category. See initializeVolumeCurves() static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][DEVICE_CATEGORY_CNT]; // descriptor for audio outputs. Used to maintain current configuration of each opened audio output // and keep track of the usage of this output by each audio stream type. class AudioOutputDescriptor: public AudioPortConfig { public: AudioOutputDescriptor(const sp& profile); status_t dump(int fd); audio_devices_t device() const; void changeRefCount(audio_stream_type_t stream, int delta); bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } audio_devices_t supportedDevices(); uint32_t latency(); bool sharesHwModuleWith(const sp outputDesc); bool isActive(uint32_t inPastMs = 0) const; bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0, nsecs_t sysTime = 0) const; bool isStrategyActive(routing_strategy strategy, uint32_t inPastMs = 0, nsecs_t sysTime = 0) const; virtual void toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig = NULL) const; virtual sp getAudioPort() const { return mProfile; } void toAudioPort(struct audio_port *port) const; audio_port_handle_t mId; audio_io_handle_t mIoHandle; // output handle uint32_t mLatency; // audio_output_flags_t mFlags; // audio_devices_t mDevice; // current device this output is routed to AudioMix *mPolicyMix; // non NULL when used by a dynamic policy audio_patch_handle_t mPatchHandle; uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output nsecs_t mStopTime[AUDIO_STREAM_CNT]; sp mOutput1; // used by duplicated outputs: first output sp mOutput2; // used by duplicated outputs: second output float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter const sp mProfile; // I/O profile this output derives from bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible // device selection. See checkDeviceMuteStrategies() uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) }; // descriptor for audio inputs. Used to maintain current configuration of each opened audio input // and keep track of the usage of this input. class AudioInputDescriptor: public AudioPortConfig { public: AudioInputDescriptor(const sp& profile); status_t dump(int fd); audio_port_handle_t mId; audio_io_handle_t mIoHandle; // input handle audio_devices_t mDevice; // current device this input is routed to AudioMix *mPolicyMix; // non NULL when used by a dynamic policy audio_patch_handle_t mPatchHandle; uint32_t mRefCount; // number of AudioRecord clients using // this input uint32_t mOpenRefCount; audio_source_t mInputSource; // input source selected by application //(mediarecorder.h) const sp mProfile; // I/O profile this output derives from SortedVector mSessions; // audio sessions attached to this input bool mIsSoundTrigger; // used by a soundtrigger capture virtual void toAudioPortConfig(struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig = NULL) const; virtual sp getAudioPort() const { return mProfile; } void toAudioPort(struct audio_port *port) const; }; // stream descriptor used for volume control class StreamDescriptor { public: StreamDescriptor(); int getVolumeIndex(audio_devices_t device); void dump(int fd); int mIndexMin; // min volume index int mIndexMax; // max volume index KeyedVector mIndexCur; // current volume index per device bool mCanBeMuted; // true is the stream can be muted const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT]; }; // stream descriptor used for volume control class EffectDescriptor : public RefBase { public: status_t dump(int fd); int mIo; // io the effect is attached to routing_strategy mStrategy; // routing strategy the effect is associated to int mSession; // audio session the effect is on effect_descriptor_t mDesc; // effect descriptor bool mEnabled; // enabled state: CPU load being used or not }; void addOutput(audio_io_handle_t output, sp outputDesc); void addInput(audio_io_handle_t input, sp inputDesc); // return the strategy corresponding to a given stream type static routing_strategy getStrategy(audio_stream_type_t stream); // return appropriate device for streams handled by the specified strategy according to current // phone state, connected devices... // if fromCache is true, the device is returned from mDeviceForStrategy[], // otherwise it is determine by current state // (device connected,phone state, force use, a2dp output...) // This allows to: // 1 speed up process when the state is stable (when starting or stopping an output) // 2 access to either current device selection (fromCache == true) or // "future" device selection (fromCache == false) when called from a context // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND // before updateDevicesAndOutputs() is called. virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy, bool fromCache); // change the route of the specified output. Returns the number of ms we have slept to // allow new routing to take effect in certain cases. uint32_t setOutputDevice(audio_io_handle_t output, audio_devices_t device, bool force = false, int delayMs = 0, audio_patch_handle_t *patchHandle = NULL, const char* address = NULL); status_t resetOutputDevice(audio_io_handle_t output, int delayMs = 0, audio_patch_handle_t *patchHandle = NULL); status_t setInputDevice(audio_io_handle_t input, audio_devices_t device, bool force = false, audio_patch_handle_t *patchHandle = NULL); status_t resetInputDevice(audio_io_handle_t input, audio_patch_handle_t *patchHandle = NULL); // select input device corresponding to requested audio source virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource, AudioMix **policyMix = NULL); // return io handle of active input or 0 if no input is active // Only considers inputs from physical devices (e.g. main mic, headset mic) when // ignoreVirtualInputs is true. audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true); uint32_t activeInputsCount() const; // initialize volume curves for each strategy and device category void initializeVolumeCurves(); // compute the actual volume for a given stream according to the requested index and a particular // device virtual float computeVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, audio_devices_t device); // check that volume change is permitted, compute and send new volume to audio hardware virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); // apply all stream volumes to the specified output and device void applyStreamVolumes(audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false); // Mute or unmute all streams handled by the specified strategy on the specified output void setStrategyMute(routing_strategy strategy, bool on, audio_io_handle_t output, int delayMs = 0, audio_devices_t device = (audio_devices_t)0); // Mute or unmute the stream on the specified output void setStreamMute(audio_stream_type_t stream, bool on, audio_io_handle_t output, int delayMs = 0, audio_devices_t device = (audio_devices_t)0); // handle special cases for sonification strategy while in call: mute streams or replace by // a special tone in the device used for communication void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange); // true if device is in a telephony or VoIP call virtual bool isInCall(); // true if given state represents a device in a telephony or VoIP call virtual bool isStateInCall(int state); // when a device is connected, checks if an open output can be routed // to this device. If none is open, tries to open one of the available outputs. // Returns an output suitable to this device or 0. // when a device is disconnected, checks if an output is not used any more and // returns its handle if any. // transfers the audio tracks and effects from one output thread to another accordingly. status_t checkOutputsForDevice(const sp devDesc, audio_policy_dev_state_t state, SortedVector& outputs, const String8 address); status_t checkInputsForDevice(audio_devices_t device, audio_policy_dev_state_t state, SortedVector& inputs, const String8 address); // close an output and its companion duplicating output. void closeOutput(audio_io_handle_t output); // close an input. void closeInput(audio_io_handle_t input); // checks and if necessary changes outputs used for all strategies. // must be called every time a condition that affects the output choice for a given strategy // changes: connected device, phone state, force use... // Must be called before updateDevicesAndOutputs() void checkOutputForStrategy(routing_strategy strategy); // Same as checkOutputForStrategy() but for a all strategies in order of priority void checkOutputForAllStrategies(); // manages A2DP output suspend/restore according to phone state and BT SCO usage void checkA2dpSuspend(); // returns the A2DP output handle if it is open or 0 otherwise audio_io_handle_t getA2dpOutput(); // selects the most appropriate device on output for current state // must be called every time a condition that affects the device choice for a given output is // changed: connected device, phone state, force use, output start, output stop.. // see getDeviceForStrategy() for the use of fromCache parameter audio_devices_t getNewOutputDevice(audio_io_handle_t output, bool fromCache); // updates cache of device used by all strategies (mDeviceForStrategy[]) // must be called every time a condition that affects the device choice for a given strategy is // changed: connected device, phone state, force use... // cached values are used by getDeviceForStrategy() if parameter fromCache is true. // Must be called after checkOutputForAllStrategies() void updateDevicesAndOutputs(); // selects the most appropriate device on input for current state audio_devices_t getNewInputDevice(audio_io_handle_t input); virtual uint32_t getMaxEffectsCpuLoad(); virtual uint32_t getMaxEffectsMemory(); #ifdef AUDIO_POLICY_TEST virtual bool threadLoop(); void exit(); int testOutputIndex(audio_io_handle_t output); #endif //AUDIO_POLICY_TEST status_t setEffectEnabled(const sp& effectDesc, bool enabled); // returns the category the device belongs to with regard to volume curve management static device_category getDeviceCategory(audio_devices_t device); // extract one device relevant for volume control from multiple device selection static audio_devices_t getDeviceForVolume(audio_devices_t device); SortedVector getOutputsForDevice(audio_devices_t device, DefaultKeyedVector > openOutputs); bool vectorsEqual(SortedVector& outputs1, SortedVector& outputs2); // mute/unmute strategies using an incompatible device combination // if muting, wait for the audio in pcm buffer to be drained before proceeding // if unmuting, unmute only after the specified delay // Returns the number of ms waited virtual uint32_t checkDeviceMuteStrategies(sp outputDesc, audio_devices_t prevDevice, uint32_t delayMs); audio_io_handle_t selectOutput(const SortedVector& outputs, audio_output_flags_t flags, audio_format_t format); // samplingRate parameter is an in/out and so may be modified sp getInputProfile(audio_devices_t device, String8 address, uint32_t& samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_input_flags_t flags); sp getProfileForDirectOutput(audio_devices_t device, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags); audio_io_handle_t selectOutputForEffects(const SortedVector& outputs); bool isNonOffloadableEffectEnabled(); status_t addAudioPatch(audio_patch_handle_t handle, const sp& patch); status_t removeAudioPatch(audio_patch_handle_t handle); sp getOutputFromId(audio_port_handle_t id) const; sp getInputFromId(audio_port_handle_t id) const; sp getModuleForDevice(audio_devices_t device) const; sp getModuleFromName(const char *name) const; audio_devices_t availablePrimaryOutputDevices(); audio_devices_t availablePrimaryInputDevices(); void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0); // // Audio policy configuration file parsing (audio_policy.conf) // static uint32_t stringToEnum(const struct StringToEnum *table, size_t size, const char *name); static const char *enumToString(const struct StringToEnum *table, size_t size, uint32_t value); static bool stringToBool(const char *value); static uint32_t parseOutputFlagNames(char *name); static uint32_t parseInputFlagNames(char *name); static audio_devices_t parseDeviceNames(char *name); void loadHwModule(cnode *root); void loadHwModules(cnode *root); void loadGlobalConfig(cnode *root, const sp& module); status_t loadAudioPolicyConfig(const char *path); void defaultAudioPolicyConfig(void); uid_t mUidCached; AudioPolicyClientInterface *mpClientInterface; // audio policy client interface audio_io_handle_t mPrimaryOutput; // primary output handle // list of descriptors for outputs currently opened DefaultKeyedVector > mOutputs; // copy of mOutputs before setDeviceConnectionState() opens new outputs // reset to mOutputs when updateDevicesAndOutputs() is called. DefaultKeyedVector > mPreviousOutputs; DefaultKeyedVector > mInputs; // list of input descriptors DeviceVector mAvailableOutputDevices; // all available output devices DeviceVector mAvailableInputDevices; // all available input devices int mPhoneState; // current phone state audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; // current forced use configuration StreamDescriptor mStreams[AUDIO_STREAM_CNT]; // stream descriptors for volume control bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected audio_devices_t mDeviceForStrategy[NUM_STRATEGIES]; float mLastVoiceVolume; // last voice volume value sent to audio HAL // Maximum CPU load allocated to audio effects in 0.1 MIPS (ARMv5TE, 0 WS memory) units static const uint32_t MAX_EFFECTS_CPU_LOAD = 1000; // Maximum memory allocated to audio effects in KB static const uint32_t MAX_EFFECTS_MEMORY = 512; uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects uint32_t mTotalEffectsMemory; // current memory used by effects KeyedVector > mEffects; // list of registered audio effects bool mA2dpSuspended; // true if A2DP output is suspended sp mDefaultOutputDevice; // output device selected by default at boot time bool mSpeakerDrcEnabled;// true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER path // to boost soft sounds, used to adjust volume curves accordingly Vector < sp > mHwModules; volatile int32_t mNextUniqueId; volatile int32_t mAudioPortGeneration; DefaultKeyedVector > mAudioPatches; DefaultKeyedVector mSoundTriggerSessions; sp mCallTxPatch; sp mCallRxPatch; // for supporting "beacon" streams, i.e. streams that only play on speaker, and never // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing enum { STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON }; uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams bool mBeaconMuted; // has STREAM_TTS been muted // custom mix entry in mPolicyMixes class AudioPolicyMix : public RefBase { public: AudioPolicyMix() {} AudioMix mMix; // Audio policy mix descriptor sp mOutput; // Corresponding output stream }; DefaultKeyedVector > mPolicyMixes; // list of registered mixes #ifdef AUDIO_POLICY_TEST Mutex mLock; Condition mWaitWorkCV; int mCurOutput; bool mDirectOutput; audio_io_handle_t mTestOutputs[NUM_TEST_OUTPUTS]; int mTestInput; uint32_t mTestDevice; uint32_t mTestSamplingRate; uint32_t mTestFormat; uint32_t mTestChannels; uint32_t mTestLatencyMs; #endif //AUDIO_POLICY_TEST static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, int indexInUi); private: // updates device caching and output for streams that can influence the // routing of notifications void handleNotificationRoutingForStream(audio_stream_type_t stream); static bool isVirtualInputDevice(audio_devices_t device); static bool deviceDistinguishesOnAddress(audio_devices_t device); // find the outputs on a given output descriptor that have the given address. // to be called on an AudioOutputDescriptor whose supported devices (as defined // in mProfile->mSupportedDevices) matches the device whose address is to be matched. // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one // where addresses are used to distinguish between one connected device and another. void findIoHandlesByAddress(sp desc /*in*/, const String8 address /*in*/, SortedVector& outputs /*out*/); uint32_t nextUniqueId(); uint32_t nextAudioPortGeneration(); uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; } // internal method to return the output handle for the given device and format audio_io_handle_t getOutputForDevice( audio_devices_t device, audio_session_t session, audio_stream_type_t stream, uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, const audio_offload_info_t *offloadInfo); // internal function to derive a stream type value from audio attributes audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr); // return true if any output is playing anything besides the stream to ignore bool isAnyOutputActive(audio_stream_type_t streamToIgnore); // event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON // returns 0 if no mute/unmute event happened, the largest latency of the device where // the mute/unmute happened uint32_t handleEventForBeacon(int event); uint32_t setBeaconMute(bool mute); bool isValidAttributes(const audio_attributes_t *paa); }; };