#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 <hardware_legacy/AudioPolicyInterface.h>
namespace android_audio_legacy {
using android::KeyedVector;
+ using android::DefaultKeyedVector;
+ using android::SortedVector;
// ----------------------------------------------------------------------------
// 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
virtual ~AudioPolicyManagerBase();
// AudioPolicyInterface
- virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
AudioSystem::device_connection_state state,
const char *device_address);
- virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+ virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
const char *device_address);
virtual void setPhoneState(int state);
- virtual void setRingerMode(uint32_t mode, uint32_t mask);
virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage);
virtual void setSystemProperty(const char* property, const char* value);
uint32_t format,
uint32_t channels,
AudioSystem::audio_in_acoustics acoustics);
+
// indicates to the audio policy manager that the input starts being used.
virtual status_t startInput(audio_io_handle_t input);
+
// indicates to the audio policy manager that the input stops being used.
virtual status_t stopInput(audio_io_handle_t input);
virtual void releaseInput(audio_io_handle_t input);
virtual void initStreamVolume(AudioSystem::stream_type stream,
int indexMin,
int indexMax);
- virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index);
- virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index);
+ virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream,
+ int *index,
+ audio_devices_t device);
// return the strategy corresponding to a given stream type
virtual uint32_t getStrategyForStream(AudioSystem::stream_type stream);
// return the enabled output devices for the given stream type
- virtual uint32_t getDevicesForStream(AudioSystem::stream_type stream);
+ virtual audio_devices_t getDevicesForStream(AudioSystem::stream_type stream);
- virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc);
- virtual status_t registerEffect(effect_descriptor_t *desc,
+ virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc);
+ 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(int stream, uint32_t inPastMs = 0) const;
+ virtual bool isSourceActive(audio_source_t source) const;
virtual status_t dump(int fd);
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_CNT
+ };
+
+ class IOProfile;
+
+ class HwModule {
+ public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ audio_module_handle_t mHandle;
+ Vector <IOProfile *> mOutputProfiles; // output profiles exposed by this module
+ Vector <IOProfile *> mInputProfiles; // input profiles exposed by this module
+ };
+
+ // 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:
+ IOProfile(HwModule *module);
+ ~IOProfile();
+
+ bool isCompatibleProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ audio_output_flags_t flags) const;
+
+ void dump(int fd);
+
+ // 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
+ audio_devices_t mSupportedDevices; // supported devices (devices this output can be
+ // routed to)
+ audio_output_flags_t mFlags; // attribute flags (e.g primary output,
+ // direct output...). For outputs only.
+ HwModule *mModule; // audio HW module exposing this I/O stream
+ };
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[AudioPolicyManagerBase::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[AudioPolicyManagerBase::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:
- AudioOutputDescriptor();
+ AudioOutputDescriptor(const IOProfile *profile);
status_t dump(int fd);
- uint32_t device();
+ audio_devices_t device();
void changeRefCount(AudioSystem::stream_type, int delta);
uint32_t refCount();
uint32_t strategyRefCount(routing_strategy strategy);
bool isUsedByStrategy(routing_strategy strategy) { return (strategyRefCount(strategy) != 0);}
- bool isDuplicated() { return (mOutput1 != NULL && mOutput2 != NULL); }
+ bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); }
+ audio_devices_t supportedDevices();
+ uint32_t latency();
+ bool sharesHwModuleWith(const AudioOutputDescriptor *outputDesc);
+ bool isActive(uint32_t inPastMs) const;
audio_io_handle_t mId; // output handle
uint32_t mSamplingRate; //
- uint32_t mFormat; //
- uint32_t mChannels; // output configuration
+ audio_format_t mFormat; //
+ audio_channel_mask_t mChannelMask; // output configuration
uint32_t mLatency; //
- AudioSystem::output_flags mFlags; //
- uint32_t mDevice; // current device this output is routed to
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
uint32_t mRefCount[AudioSystem::NUM_STREAM_TYPES]; // number of streams of each type using this output
nsecs_t mStopTime[AudioSystem::NUM_STREAM_TYPES];
AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output
AudioOutputDescriptor *mOutput2; // used by duplicated outputs: second output
float mCurVolume[AudioSystem::NUM_STREAM_TYPES]; // current stream volume
int mMuteCount[AudioSystem::NUM_STREAM_TYPES]; // mute request counter
+ const IOProfile *mProfile; // I/O profile this output derives from
+ bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible
+ // device selection. See checkDeviceMuteStrategies()
};
// descriptor for audio inputs. Used to maintain current configuration of each opened audio input
class AudioInputDescriptor
{
public:
- AudioInputDescriptor();
+ AudioInputDescriptor(const IOProfile *profile);
status_t dump(int fd);
uint32_t mSamplingRate; //
- uint32_t mFormat; // input configuration
- uint32_t mChannels; //
- AudioSystem::audio_in_acoustics mAcoustics; //
- uint32_t mDevice; // current device this input is routed to
+ audio_format_t mFormat; // input configuration
+ audio_channel_mask_t mChannelMask; //
+ audio_devices_t mDevice; // current device this input is routed to
uint32_t mRefCount; // number of AudioRecord clients using this output
- int mInputSource; // input source selected by application (mediarecorder.h)
+ int mInputSource; // input source selected by application (mediarecorder.h)
+ const IOProfile *mProfile; // I/O profile this output derives from
};
// stream descriptor used for volume control
class StreamDescriptor
{
public:
- StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mIndexCur(1), mCanBeMuted(true) {}
+ StreamDescriptor();
- void dump(char* buffer, size_t size);
+ int getVolumeIndex(audio_devices_t device);
+ void dump(int fd);
int mIndexMin; // min volume index
int mIndexMax; // max volume index
- int mIndexCur; // current volume index
+ KeyedVector<audio_devices_t, int> mIndexCur; // current volume index per device
bool mCanBeMuted; // true is the stream can be muted
- // 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.
- int mVolIndex[NUM_VOL_CURVE_KNEES+2]; // minimum index, index at knees, and max index
- float mVolDbAtt[NUM_VOL_CURVE_KNEES+2]; // largest attenuation, attenuation at knees,
- // and attenuation at max vol (usually 0dB)
- // indices in mVolIndex and mVolDbAtt respectively for points at lowest volume, knee 1,
- // knee 2 and highest volume.
- enum { VOLMIN = 0, VOLKNEE1 = 1, VOLKNEE2 = 2, VOLMAX = 3 };
+ const VolumeCurvePoint *mVolumeCurve[DEVICE_CATEGORY_CNT];
};
// stream descriptor used for volume control
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 id, AudioOutputDescriptor *outputDesc);
// return the strategy corresponding to a given stream type
static routing_strategy getStrategy(AudioSystem::stream_type 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 determined
- // by current state (device connected, phone state, force use, a2dp output...)
+ // 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 updateDeviceForStrategy() is called.
- virtual uint32_t getDeviceForStrategy(routing_strategy strategy, bool fromCache = true);
- // change the route of the specified output
- void setOutputDevice(audio_io_handle_t output, uint32_t device, bool force = false, int delayMs = 0);
+ // 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);
+
// select input device corresponding to requested audio source
- virtual uint32_t getDeviceForInputSource(int inputSource);
+ virtual audio_devices_t getDeviceForInputSource(int inputSource);
+
// return io handle of active input or 0 if no input is active
- audio_io_handle_t getActiveInput();
- virtual void initializeVolumeCurves();
+ // Only considers inputs from physical devices (e.g. main mic, headset mic) when
+ // ignoreVirtualInputs is true.
+ audio_io_handle_t getActiveInput(bool ignoreVirtualInputs = true);
+
+ // 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(int stream, int index, audio_io_handle_t output, uint32_t device);
+ virtual float computeVolume(int 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(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs = 0, bool force = false);
+ status_t checkAndSetVolume(int 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, uint32_t device, int delayMs = 0, bool force = false);
+ 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);
+ 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(int stream, bool on, audio_io_handle_t output, int delayMs = 0);
+ void setStreamMute(int 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(int stream, bool starting, bool stateChange);
- // true is current platform implements a back microphone
- virtual bool hasBackMicrophone() const { return false; }
+
// 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);
-#ifdef WITH_A2DP
- // true is current platform supports suplication of notifications and ringtones over A2DP output
- virtual bool a2dpUsedForSonification() const { return true; }
- status_t handleA2dpConnection(AudioSystem::audio_devices device,
- const char *device_address);
- status_t handleA2dpDisconnection(AudioSystem::audio_devices device,
- const char *device_address);
- void closeA2dpOutputs();
- // checks and if necessary changes output (a2dp, duplicated or hardware) used for all strategies.
- // must be called every time a condition that affects the output choice for a given strategy is
- // changed: connected device, phone state, force use...
- // Must be called before updateDeviceForStrategy()
+ // 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(audio_devices_t device,
+ AudioSystem::device_connection_state state,
+ SortedVector<audio_io_handle_t>& outputs);
+
+ // close an output and its companion duplicating output.
+ void closeOutput(audio_io_handle_t output);
+
+ // 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();
-#endif
+
+ // 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
- uint32_t getNewDevice(audio_io_handle_t output, bool fromCache = true);
+
+ audio_devices_t getNewDevice(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 updateDeviceForStrategy();
+
+ void updateDevicesAndOutputs();
+
// true if current platform requires a specific output to be opened for this particular
// set of parameters. This function is called by getOutput() and is implemented by platform
// specific audio policy manager.
- virtual bool needsDirectOuput(AudioSystem::stream_type stream,
- uint32_t samplingRate,
- uint32_t format,
- uint32_t channels,
- AudioSystem::output_flags flags,
- uint32_t device);
+ virtual bool needsDirectOuput(audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ audio_devices_t device);
+
virtual uint32_t getMaxEffectsCpuLoad();
virtual uint32_t getMaxEffectsMemory();
#ifdef AUDIO_POLICY_TEST
int testOutputIndex(audio_io_handle_t output);
#endif //AUDIO_POLICY_TEST
+ status_t setEffectEnabled(EffectDescriptor *pDesc, 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, 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(AudioOutputDescriptor *outputDesc,
+ audio_devices_t prevDevice,
+ uint32_t delayMs);
+
+ audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
+ AudioSystem::output_flags flags);
+ IOProfile *getInputProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask);
+ IOProfile *getProfileForDirectOutput(audio_devices_t device,
+ uint32_t samplingRate,
+ uint32_t format,
+ uint32_t channelMask,
+ audio_output_flags_t flags);
+ //
+ // Audio policy configuration file parsing (audio_policy.conf)
+ //
+ static uint32_t stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name);
+ static audio_output_flags_t parseFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+ void loadSamplingRates(char *name, IOProfile *profile);
+ void loadFormats(char *name, IOProfile *profile);
+ void loadOutChannels(char *name, IOProfile *profile);
+ void loadInChannels(char *name, IOProfile *profile);
+ status_t loadOutput(cnode *root, HwModule *module);
+ status_t loadInput(cnode *root, HwModule *module);
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root);
+ status_t loadAudioPolicyConfig(const char *path);
+ void defaultAudioPolicyConfig(void);
+
+
AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
- audio_io_handle_t mHardwareOutput; // hardware output handler
- audio_io_handle_t mA2dpOutput; // A2DP output handler
- audio_io_handle_t mDuplicatedOutput; // duplicated output handler: outputs to hardware and A2DP.
-
- KeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs; // list of output descriptors
- KeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
- uint32_t mAvailableOutputDevices; // bit field of all available output devices
- uint32_t mAvailableInputDevices; // bit field of all available input devices
+ audio_io_handle_t mPrimaryOutput; // primary output handle
+ // list of descriptors for outputs currently opened
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs; // list of input descriptors
+ audio_devices_t mAvailableOutputDevices; // bit field of all available output devices
+ audio_devices_t mAvailableInputDevices; // bit field of all available input devices
+ // without AUDIO_DEVICE_BIT_IN to allow direct bit
+ // field comparisons
int mPhoneState; // current phone state
- uint32_t mRingerMode; // current ringer mode
AudioSystem::forced_config mForceUse[AudioSystem::NUM_FORCE_USE]; // current forced use configuration
StreamDescriptor mStreams[AudioSystem::NUM_STREAM_TYPES]; // stream descriptors for volume control
String8 mA2dpDeviceAddress; // A2DP device MAC address
String8 mScoDeviceAddress; // SCO device MAC address
+ String8 mUsbCardAndDevice; // USB audio ALSA card and device numbers:
+ // card=<card_number>;device=<><device_number>
bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
- uint32_t mDeviceForStrategy[NUM_STRATEGIES];
+ 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
uint32_t mTotalEffectsMemory; // current memory used by effects
KeyedVector<int, EffectDescriptor *> mEffects; // list of registered audio effects
bool mA2dpSuspended; // true if A2DP output is suspended
+ bool mHasA2dp; // true on platforms with support for bluetooth A2DP
+ bool mHasUsb; // true on platforms with support for USB audio
+ bool mHasRemoteSubmix; // true on platforms with support for remote presentation of a submix
+ audio_devices_t mAttachedOutputDevices; // output devices always available on the platform
+ audio_devices_t mDefaultOutputDevice; // output device selected by default at boot time
+ // (must be in mAttachedOutputDevices)
+
+ Vector <HwModule *> mHwModules;
#ifdef AUDIO_POLICY_TEST
Mutex mLock;
#endif //AUDIO_POLICY_TEST
private:
- static float volIndexToAmpl(uint32_t device, const StreamDescriptor& streamDesc,
+ 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(AudioSystem::stream_type stream);
+ static bool isVirtualInputDevice(audio_devices_t device);
};
};
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