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Diffstat (limited to 'services/audiopolicy/AudioPolicyManager.h')
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diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h new file mode 100644 index 0000000..0ea7b97 --- /dev/null +++ b/services/audiopolicy/AudioPolicyManager.h @@ -0,0 +1,856 @@ +/* + * 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 <stdint.h> +#include <sys/types.h> +#include <cutils/config_utils.h> +#include <cutils/misc.h> +#include <utils/Timers.h> +#include <utils/Errors.h> +#include <utils/KeyedVector.h> +#include <utils/SortedVector.h> +#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 audio_io_handle_t getOutputForAttr(const audio_attributes_t *attr, + 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, + int session = 0); + virtual status_t stopOutput(audio_io_handle_t output, + audio_stream_type_t stream, + int session = 0); + virtual void releaseOutput(audio_io_handle_t output); + virtual audio_io_handle_t getInput(audio_source_t inputSource, + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_session_t session, + 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 + 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); + +protected: + + enum routing_strategy { + STRATEGY_MEDIA, + STRATEGY_PHONE, + STRATEGY_SONIFICATION, + STRATEGY_SONIFICATION_RESPECTFUL, + STRATEGY_DTMF, + STRATEGY_ENFORCED_AUDIBLE, + 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<HwModule>& module); + virtual ~AudioPort() {} + + virtual void toAudioPort(struct audio_port *port) const; + + void importAudioPort(const sp<AudioPort> 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 <uint32_t> mSamplingRates; // supported sampling rates + Vector <audio_channel_mask_t> mChannelMasks; // supported channel masks + Vector <audio_format_t> mFormats; // supported audio formats + Vector < sp<AudioGain> > mGains; // gain controllers + sp<HwModule> 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<AudioPort> 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<DeviceDescriptor>& other) const; + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + virtual sp<AudioPort> 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<DeviceDescriptor> > + { + public: + DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} + + ssize_t add(const sp<DeviceDescriptor>& item); + ssize_t remove(const sp<DeviceDescriptor>& item); + ssize_t indexOf(const sp<DeviceDescriptor>& item) const; + + audio_devices_t types() const { return mDeviceTypes; } + + void loadDevicesFromType(audio_devices_t types); + void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); + + sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const; + DeviceVector getDevicesFromType(audio_devices_t types) const; + sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const; + sp<DeviceDescriptor> 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<HwModule>& 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, + 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); + + 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<IOProfile> > mOutputProfiles; // output profiles exposed by this module + Vector < sp<IOProfile> > 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]; + // 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<IOProfile>& 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<AudioOutputDescriptor> 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<AudioPort> 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 + 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<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output + sp<AudioOutputDescriptor> 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<IOProfile> 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<IOProfile>& 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 + 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<IOProfile> mProfile; // I/O profile this output derives from + SortedVector<audio_session_t> 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<AudioPort> 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<audio_devices_t, int> 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<AudioOutputDescriptor> outputDesc); + void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> 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); + + // 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 + 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<DeviceDescriptor> devDesc, + audio_policy_dev_state_t state, + SortedVector<audio_io_handle_t>& outputs, + const String8 address); + + status_t checkInputsForDevice(audio_devices_t device, + audio_policy_dev_state_t state, + SortedVector<audio_io_handle_t>& 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<EffectDescriptor>& 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<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, + DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs); + bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, + SortedVector<audio_io_handle_t>& 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 + uint32_t checkDeviceMuteStrategies(sp<AudioOutputDescriptor> outputDesc, + audio_devices_t prevDevice, + uint32_t delayMs); + + audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs, + audio_output_flags_t flags, + audio_format_t format); + // samplingRate parameter is an in/out and so may be modified + sp<IOProfile> getInputProfile(audio_devices_t device, + uint32_t& samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags); + sp<IOProfile> 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<audio_io_handle_t>& outputs); + + bool isNonOffloadableEffectEnabled(); + + status_t addAudioPatch(audio_patch_handle_t handle, + const sp<AudioPatch>& patch); + status_t removeAudioPatch(audio_patch_handle_t handle); + + sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const; + sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const; + sp<HwModule> getModuleForDevice(audio_devices_t device) const; + sp<HwModule> 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<HwModule>& 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<audio_io_handle_t, sp<AudioOutputDescriptor> > mOutputs; + // copy of mOutputs before setDeviceConnectionState() opens new outputs + // reset to mOutputs when updateDevicesAndOutputs() is called. + DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs; + DefaultKeyedVector<audio_io_handle_t, sp<AudioInputDescriptor> > 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<int, sp<EffectDescriptor> > mEffects; // list of registered audio effects + bool mA2dpSuspended; // true if A2DP output is suspended + sp<DeviceDescriptor> 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<HwModule> > mHwModules; + volatile int32_t mNextUniqueId; + volatile int32_t mAudioPortGeneration; + + DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches; + + DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions; + + sp<AudioPatch> mCallTxPatch; + sp<AudioPatch> mCallRxPatch; + +#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 + +private: + static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc, + int indexInUi); + // 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<AudioOutputDescriptor> desc /*in*/, + const String8 address /*in*/, + SortedVector<audio_io_handle_t>& 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_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); +}; + +}; |