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* 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
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* See the License for the specific language governing permissions and
* limitations under the License.
#pragma once
#include <atomic>
#include <functional>
#include <memory>
#include <unordered_set>
#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 <media/AudioParameter.h>
#include <media/AudioPolicy.h>
#include <media/PatchBuilder.h>
#include "AudioPolicyInterface.h"
#include <AudioPolicyManagerInterface.h>
#include <AudioPolicyManagerObserver.h>
#include <AudioGain.h>
#include <AudioPolicyConfig.h>
#include <AudioPort.h>
#include <AudioPatch.h>
#include <AudioProfile.h>
#include <DeviceDescriptor.h>
#include <IOProfile.h>
#include <HwModule.h>
#include <AudioInputDescriptor.h>
#include <AudioOutputDescriptor.h>
#include <AudioPolicyMix.h>
#include <EffectDescriptor.h>
#include <SoundTriggerSession.h>
#include "TypeConverter.h"
namespace android {
// ----------------------------------------------------------------------------
// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
// Max volume difference on A2DP between playing media and STRATEGY_SONIFICATION streams: 12dB
// Time in milliseconds during which we consider that music is still active after a music
// track was stopped - see computeVolume()
// Time in milliseconds during witch some streams are muted while the audio path
// is switched
#define MUTE_TIME_MS 2000
// multiplication factor applied to output latency when calculating a safe mute delay when
// invalidating tracks
// Default minimum length allowed for offloading a compressed track
// Can be overridden by the audio.offload.min.duration.secs property
// ----------------------------------------------------------------------------
// AudioPolicyManager implements audio policy manager behavior common to all platforms.
// ----------------------------------------------------------------------------
class AudioPolicyManager : public AudioPolicyInterface, public AudioPolicyManagerObserver
explicit AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
virtual ~AudioPolicyManager();
// AudioPolicyInterface
virtual status_t setDeviceConnectionState(audio_devices_t device,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat);
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address);
virtual status_t handleDeviceConfigChange(audio_devices_t device,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat);
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);
status_t getOutputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *output,
audio_session_t session,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
audio_port_handle_t *portId,
std::vector<audio_io_handle_t> *secondaryOutputs) override;
virtual status_t startOutput(audio_port_handle_t portId);
virtual status_t stopOutput(audio_port_handle_t portId);
virtual void releaseOutput(audio_port_handle_t portId);
virtual status_t getInputForAttr(const audio_attributes_t *attr,
audio_io_handle_t *input,
audio_unique_id_t riid,
audio_session_t session,
uid_t uid,
const audio_config_base_t *config,
audio_input_flags_t flags,
audio_port_handle_t *selectedDeviceId,
input_type_t *inputType,
audio_port_handle_t *portId);
// indicates to the audio policy manager that the input starts being used.
virtual status_t startInput(audio_port_handle_t portId);
// indicates to the audio policy manager that the input stops being used.
virtual status_t stopInput(audio_port_handle_t portId);
virtual void releaseInput(audio_port_handle_t portId);
virtual void checkCloseInputs();
* @brief initStreamVolume: even if the engine volume files provides min and max, keep this
* api for compatibility reason.
* AudioServer will get the min and max and may overwrite them if:
* -using property (highest priority)
* -not defined (-1 by convention), case when still using apm volume tables XML files
* @param stream to be considered
* @param indexMin to set
* @param indexMax to set
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);
virtual status_t setVolumeIndexForAttributes(const audio_attributes_t &attr,
int index,
audio_devices_t device);
virtual status_t getVolumeIndexForAttributes(const audio_attributes_t &attr,
int &index,
audio_devices_t device);
virtual status_t getMaxVolumeIndexForAttributes(const audio_attributes_t &attr, int &index);
virtual status_t getMinVolumeIndexForAttributes(const audio_attributes_t &attr, int &index);
status_t setVolumeCurveIndex(int index,
audio_devices_t device,
IVolumeCurves &volumeCurves);
status_t getVolumeIndex(const IVolumeCurves &curves, int &index,
audio_devices_t device) const;
// return the strategy corresponding to a given stream type
virtual uint32_t getStrategyForStream(audio_stream_type_t stream)
return streamToStrategy(stream);
product_strategy_t streamToStrategy(audio_stream_type_t stream) const
auto attributes = mEngine->getAttributesForStreamType(stream);
return mEngine->getProductStrategyForAttributes(attributes);
// 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);
status_t moveEffectsToIo(const std::vector<int>& ids, audio_io_handle_t io) override;
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;
// helpers for dump(int fd)
void dumpManualSurroundFormats(String8 *dst) const;
void dump(String8 *dst) const;
status_t dump(int fd) override;
status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy) override;
virtual bool isOffloadSupported(const audio_offload_info_t& offloadInfo);
virtual bool isDirectOutputSupported(const audio_config_base_t& config,
const audio_attributes_t& attributes);
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 releaseResourcesForUid(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)
return mSoundTriggerSessions.releaseSession(session);
virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes);
virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
virtual status_t setUidDeviceAffinities(uid_t uid,
const Vector<AudioDeviceTypeAddr>& devices);
virtual status_t removeUidDeviceAffinities(uid_t uid);
virtual status_t startAudioSource(const struct audio_port_config *source,
const audio_attributes_t *attributes,
audio_port_handle_t *portId,
uid_t uid);
virtual status_t stopAudioSource(audio_port_handle_t portId);
virtual status_t setMasterMono(bool mono);
virtual status_t getMasterMono(bool *mono);
virtual float getStreamVolumeDB(
audio_stream_type_t stream, int index, audio_devices_t device);
virtual status_t getSurroundFormats(unsigned int *numSurroundFormats,
audio_format_t *surroundFormats,
bool *surroundFormatsEnabled,
bool reported);
virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
std::vector<audio_format_t> *formats);
virtual void setAppState(uid_t uid, app_state_t state);
virtual bool isHapticPlaybackSupported();
virtual status_t listAudioProductStrategies(AudioProductStrategyVector &strategies)
return mEngine->listAudioProductStrategies(strategies);
virtual status_t getProductStrategyFromAudioAttributes(const AudioAttributes &aa,
product_strategy_t &productStrategy)
productStrategy = mEngine->getProductStrategyForAttributes(aa.getAttributes());
return productStrategy != PRODUCT_STRATEGY_NONE ? NO_ERROR : BAD_VALUE;
virtual status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups)
return mEngine->listAudioVolumeGroups(groups);
virtual status_t getVolumeGroupFromAudioAttributes(const AudioAttributes &aa,
volume_group_t &volumeGroup)
volumeGroup = mEngine->getVolumeGroupForAttributes(aa.getAttributes());
return volumeGroup != VOLUME_GROUP_NONE ? NO_ERROR : BAD_VALUE;
// A constructor that allows more fine-grained control over initialization process,
// used in automatic tests.
AudioPolicyManager(AudioPolicyClientInterface *clientInterface, bool forTesting);
// These methods should be used when finer control over APM initialization
// is needed, e.g. in tests. Must be used in conjunction with the constructor
// that only performs fields initialization. The public constructor comprises
// these steps in the following sequence:
// - field initializing constructor;
// - loadConfig;
// - initialize.
AudioPolicyConfig& getConfig() { return mConfig; }
void loadConfig();
status_t initialize();
// From AudioPolicyManagerObserver
virtual const AudioPatchCollection &getAudioPatches() const
return mAudioPatches;
virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const
return mSoundTriggerSessions;
virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const
return mPolicyMixes;
virtual const SwAudioOutputCollection &getOutputs() const
return mOutputs;
virtual const AudioInputCollection &getInputs() const
return mInputs;
virtual const DeviceVector getAvailableOutputDevices() const
return mAvailableOutputDevices;
virtual const DeviceVector getAvailableInputDevices() const
// legacy and non-legacy remote-submix are managed by the engine, do not filter
return mAvailableInputDevices;
virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const
return mDefaultOutputDevice;
std::vector<volume_group_t> getVolumeGroups() const
return mEngine->getVolumeGroups();
VolumeSource toVolumeSource(volume_group_t volumeGroup) const
return static_cast<VolumeSource>(volumeGroup);
VolumeSource toVolumeSource(const audio_attributes_t &attributes) const
return toVolumeSource(mEngine->getVolumeGroupForAttributes(attributes));
VolumeSource toVolumeSource(audio_stream_type_t stream) const
return toVolumeSource(mEngine->getVolumeGroupForStreamType(stream));
IVolumeCurves &getVolumeCurves(VolumeSource volumeSource)
auto *curves = mEngine->getVolumeCurvesForVolumeGroup(
ALOG_ASSERT(curves != nullptr, "No curves for volume source %d", volumeSource);
return *curves;
IVolumeCurves &getVolumeCurves(const audio_attributes_t &attr)
auto *curves = mEngine->getVolumeCurvesForAttributes(attr);
ALOG_ASSERT(curves != nullptr, "No curves for attributes %s", toString(attr).c_str());
return *curves;
IVolumeCurves &getVolumeCurves(audio_stream_type_t stream)
auto *curves = mEngine->getVolumeCurvesForStreamType(stream);
ALOG_ASSERT(curves != nullptr, "No curves for stream %s", toString(stream).c_str());
return *curves;
void addOutput(audio_io_handle_t output, const sp<SwAudioOutputDescriptor>& outputDesc);
void removeOutput(audio_io_handle_t output);
void addInput(audio_io_handle_t input, const sp<AudioInputDescriptor>& inputDesc);
// 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 setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
const DeviceVector &device,
bool force = false,
int delayMs = 0,
audio_patch_handle_t *patchHandle = NULL,
bool requiresMuteCheck = true);
status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
audio_patch_handle_t *patchHandle = NULL);
status_t setInputDevice(audio_io_handle_t input,
const sp<DeviceDescriptor> &device,
bool force = false,
audio_patch_handle_t *patchHandle = NULL);
status_t resetInputDevice(audio_io_handle_t input,
audio_patch_handle_t *patchHandle = NULL);
// compute the actual volume for a given stream according to the requested index and a particular
// device
virtual float computeVolume(IVolumeCurves &curves,
VolumeSource volumeSource,
int index,
audio_devices_t device);
// rescale volume index from srcStream within range of dstStream
int rescaleVolumeIndex(int srcIndex,
VolumeSource fromVolumeSource,
VolumeSource toVolumeSource);
// check that volume change is permitted, compute and send new volume to audio hardware
virtual status_t checkAndSetVolume(IVolumeCurves &curves,
VolumeSource volumeSource, int index,
const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device,
int delayMs = 0, bool force = false);
// apply all stream volumes to the specified output and device
void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
audio_devices_t device, int delayMs = 0, bool force = false);
* @brief setStrategyMute Mute or unmute all active clients on the considered output
* following the given strategy.
* @param strategy to be considered
* @param on true for mute, false for unmute
* @param outputDesc to be considered
* @param delayMs
* @param device
void setStrategyMute(product_strategy_t strategy,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
audio_devices_t device = AUDIO_DEVICE_NONE);
* @brief setVolumeSourceMute Mute or unmute the volume source on the specified output
* @param volumeSource to be muted/unmute (may host legacy streams or by extension set of
* audio attributes)
* @param on true to mute, false to umute
* @param outputDesc on which the client following the volume group shall be muted/umuted
* @param delayMs
* @param device
void setVolumeSourceMute(VolumeSource volumeSource,
bool on,
const sp<AudioOutputDescriptor>& outputDesc,
int delayMs = 0,
audio_devices_t device = AUDIO_DEVICE_NONE);
audio_mode_t getPhoneState();
// 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>& device,
audio_policy_dev_state_t state,
SortedVector<audio_io_handle_t>& outputs);
status_t checkInputsForDevice(const sp<DeviceDescriptor>& device,
audio_policy_dev_state_t state);
// 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);
// runs all the checks required for accomodating changes in devices and outputs
// if 'onOutputsChecked' callback is provided, it is executed after the outputs
// check via 'checkOutputForAllStrategies'. If the callback returns 'true',
// A2DP suspend status is rechecked.
void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked = nullptr);
* @brief checkOutputForAttributes checks and if necessary changes outputs used for the
* given audio attributes.
* must be called every time a condition that affects the output choice for a given
* attributes changes: connected device, phone state, force use...
* Must be called before updateDevicesAndOutputs()
* @param attr to be considered
void checkOutputForAttributes(const audio_attributes_t &attr);
bool followsSameRouting(const audio_attributes_t &lAttr,
const audio_attributes_t &rAttr) const;
* @brief checkOutputForAllStrategies Same as @see checkOutputForAttributes()
* but for a all product strategies in order of priority
void checkOutputForAllStrategies();
// Same as checkOutputForStrategy but for secondary outputs. Make sure if a secondary
// output condition changes, the track is properly rerouted
void checkSecondaryOutputs();
// manages A2DP output suspend/restore according to phone state and BT SCO usage
void checkA2dpSuspend();
// 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
DeviceVector getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
bool fromCache);
* @brief updateDevicesAndOutputs: updates cache of devices of the engine
* must be called every time a condition that affects the device choice is changed:
* connected device, phone state, force use...
* cached values are used by getOutputDevicesForStream()/getDevicesForAttributes if
* parameter fromCache is true.
* Must be called after checkOutputForAllStrategies()
void updateDevicesAndOutputs();
// selects the most appropriate device on input for current state
sp<DeviceDescriptor> getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
virtual uint32_t getMaxEffectsCpuLoad()
return mEffects.getMaxEffectsCpuLoad();
virtual uint32_t getMaxEffectsMemory()
return mEffects.getMaxEffectsMemory();
SortedVector<audio_io_handle_t> getOutputsForDevices(
const DeviceVector &devices, const SwAudioOutputCollection& openOutputs);
* @brief checkDeviceMuteStrategies 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
* @param outputDesc
* @param prevDevice
* @param delayMs
* @return the number of ms waited
virtual uint32_t checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
const DeviceVector &prevDevices,
uint32_t delayMs);
audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
audio_format_t format = AUDIO_FORMAT_INVALID,
audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE,
uint32_t samplingRate = 0);
// samplingRate, format, channelMask are in/out and so may be modified
sp<IOProfile> getInputProfile(const sp<DeviceDescriptor> & device,
uint32_t& samplingRate,
audio_format_t& format,
audio_channel_mask_t& channelMask,
audio_input_flags_t flags);
* @brief getProfileForOutput
* @param devices vector of descriptors, may be empty if ignoring the device is required
* @param samplingRate
* @param format
* @param channelMask
* @param flags
* @param directOnly
* @return IOProfile to be used if found, nullptr otherwise
sp<IOProfile> getProfileForOutput(const DeviceVector &devices,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags,
bool directOnly);
audio_io_handle_t selectOutputForMusicEffects();
virtual status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch)
return mAudioPatches.addAudioPatch(handle, patch);
virtual status_t removeAudioPatch(audio_patch_handle_t handle)
return mAudioPatches.removeAudioPatch(handle);
bool isPrimaryModule(const sp<HwModule> &module) const
if (module == 0 || !hasPrimaryOutput()) {
return false;
return module->getHandle() == mPrimaryOutput->getModuleHandle();
DeviceVector availablePrimaryOutputDevices() const
if (!hasPrimaryOutput()) {
return DeviceVector();
return mAvailableOutputDevices.filter(mPrimaryOutput->supportedDevices());
DeviceVector availablePrimaryModuleInputDevices() const
if (!hasPrimaryOutput()) {
return DeviceVector();
return mAvailableInputDevices.getDevicesFromHwModule(
* @brief getFirstDeviceId of the Device Vector
* @return if the collection is not empty, it returns the first device Id,
audio_port_handle_t getFirstDeviceId(const DeviceVector &devices) const
return (devices.size() > 0) ? devices.itemAt(0)->getId() : AUDIO_PORT_HANDLE_NONE;
String8 getFirstDeviceAddress(const DeviceVector &devices) const
return (devices.size() > 0) ? devices.itemAt(0)->address() : String8("");
uint32_t updateCallRouting(const DeviceVector &rxDevices, uint32_t delayMs = 0);
sp<AudioPatch> createTelephonyPatch(bool isRx, const sp<DeviceDescriptor> &device,
uint32_t delayMs);
sp<DeviceDescriptor> findDevice(
const DeviceVector& devices, audio_devices_t device) const;
audio_devices_t getModuleDeviceTypes(
const DeviceVector& devices, const char *moduleId) const;
bool isDeviceOfModule(const sp<DeviceDescriptor>& devDesc, const char *moduleId) const;
status_t startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client,
uint32_t *delayMs);
status_t stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
const sp<TrackClientDescriptor>& client);
void clearAudioPatches(uid_t uid);
void clearSessionRoutes(uid_t uid);
* @brief checkStrategyRoute: when an output is beeing rerouted, reconsider each output
* that may host a strategy playing on the considered output.
* @param ps product strategy that initiated the rerouting
* @param ouptutToSkip output that initiated the rerouting
void checkStrategyRoute(product_strategy_t ps, audio_io_handle_t ouptutToSkip);
status_t hasPrimaryOutput() const { return mPrimaryOutput != 0; }
status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
sp<SourceClientDescriptor> getSourceForAttributesOnOutput(audio_io_handle_t output,
const audio_attributes_t &attr);
void cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc);
void clearAudioSources(uid_t uid);
static bool streamsMatchForvolume(audio_stream_type_t stream1,
audio_stream_type_t stream2);
void closeActiveClients(const sp<AudioInputDescriptor>& input);
void closeClient(audio_port_handle_t portId);
const uid_t mUidCached; // AID_AUDIOSERVER
AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
sp<SwAudioOutputDescriptor> mPrimaryOutput; // primary output descriptor
// list of descriptors for outputs currently opened
SwAudioOutputCollection mOutputs;
// copy of mOutputs before setDeviceConnectionState() opens new outputs
// reset to mOutputs when updateDevicesAndOutputs() is called.
SwAudioOutputCollection mPreviousOutputs;
AudioInputCollection mInputs; // list of input descriptors
DeviceVector mAvailableOutputDevices; // all available output devices
DeviceVector mAvailableInputDevices; // all available input devices
bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
float mLastVoiceVolume; // last voice volume value sent to audio HAL
bool mA2dpSuspended; // true if A2DP output is suspended
EffectDescriptorCollection mEffects; // list of registered audio effects
sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
HwModuleCollection mHwModules; // contains only modules that have been loaded successfully
HwModuleCollection mHwModulesAll; // normally not needed, used during construction and for
// dumps
AudioPolicyConfig mConfig;
std::atomic<uint32_t> mAudioPortGeneration;
AudioPatchCollection mAudioPatches;
SoundTriggerSessionCollection mSoundTriggerSessions;
sp<AudioPatch> mCallTxPatch;
sp<AudioPatch> mCallRxPatch;
HwAudioOutputCollection mHwOutputs;
SourceClientCollection mAudioSources;
// 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 {
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
bool mTtsOutputAvailable; // true if a dedicated output for TTS stream is available
bool mMasterMono; // true if we wish to force all outputs to mono
AudioPolicyMixCollection mPolicyMixes; // list of registered mixes
audio_io_handle_t mMusicEffectOutput; // output selected for music effects
uint32_t nextAudioPortGeneration();
// Audio Policy Engine Interface.
AudioPolicyManagerInterface *mEngine;
// Surround formats that are enabled manually. Taken into account when
// "encoded surround" is forced into "manual" mode.
std::unordered_set<audio_format_t> mManualSurroundFormats;
std::unordered_map<uid_t, audio_flags_mask_t> mAllowedCapturePolicies;
// Add or remove AC3 DTS encodings based on user preferences.
void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
void modifySurroundChannelMasks(ChannelsVector *channelMasksPtr);
// Support for Multi-Stream Decoder (MSD) module
sp<DeviceDescriptor> getMsdAudioInDevice() const;
DeviceVector getMsdAudioOutDevices() const;
const AudioPatchCollection getMsdPatches() const;
status_t getBestMsdAudioProfileFor(const sp<DeviceDescriptor> &outputDevice,
bool hwAvSync,
audio_port_config *sourceConfig,
audio_port_config *sinkConfig) const;
PatchBuilder buildMsdPatch(const sp<DeviceDescriptor> &outputDevice) const;
status_t setMsdPatch(const sp<DeviceDescriptor> &outputDevice = nullptr);
// If any, resolve any "dynamic" fields of an Audio Profiles collection
void updateAudioProfiles(const sp<DeviceDescriptor>& devDesc, audio_io_handle_t ioHandle,
AudioProfileVector &profiles);
// Notify the policy client of any change of device state with AUDIO_IO_HANDLE_NONE,
// so that the client interprets it as global to audio hardware interfaces.
// It can give a chance to HAL implementer to retrieve dynamic capabilities associated
// to this device for example.
// TODO avoid opening stream to retrieve capabilities of a profile.
void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
audio_policy_dev_state_t state);
// updates device caching and output for streams that can influence the
// routing of notifications
void handleNotificationRoutingForStream(audio_stream_type_t stream);
uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
// internal method, get audio_attributes_t from either a source audio_attributes_t
// or audio_stream_type_t, respectively.
status_t getAudioAttributes(audio_attributes_t *dstAttr,
const audio_attributes_t *srcAttr,
audio_stream_type_t srcStream);
// internal method, called by getOutputForAttr() and connectAudioSource.
status_t getOutputForAttrInt(audio_attributes_t *resultAttr,
audio_io_handle_t *output,
audio_session_t session,
const audio_attributes_t *attr,
audio_stream_type_t *stream,
uid_t uid,
const audio_config_t *config,
audio_output_flags_t *flags,
audio_port_handle_t *selectedDeviceId,
bool *isRequestedDeviceForExclusiveUse,
std::vector<sp<SwAudioOutputDescriptor>> *secondaryDescs);
// internal method to return the output handle for the given device and format
audio_io_handle_t getOutputForDevices(
const DeviceVector &devices,
audio_session_t session,
audio_stream_type_t stream,
const audio_config_t *config,
audio_output_flags_t *flags,
bool forceMutingHaptic = false);
* @brief getInputForDevice selects an input handle for a given input device and
* requester context
* @param device to be used by requester, selected by policy mix rules or engine
* @param session requester session id
* @param uid requester uid
* @param attributes requester audio attributes (e.g. input source and tags matter)
* @param config requester audio configuration (e.g. sample rate, format, channel mask).
* @param flags requester input flags
* @param policyMix may be null, policy rules to be followed by the requester
* @return input io handle aka unique input identifier selected for this device.
audio_io_handle_t getInputForDevice(const sp<DeviceDescriptor> &device,
audio_session_t session,
const audio_attributes_t &attributes,
const audio_config_base_t *config,
audio_input_flags_t flags,
const sp<AudioPolicyMix> &policyMix);
// 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);
// Called by setDeviceConnectionState().
status_t setDeviceConnectionStateInt(audio_devices_t deviceType,
audio_policy_dev_state_t state,
const char *device_address,
const char *device_name,
audio_format_t encodedFormat);
void setEngineDeviceConnectionState(const sp<DeviceDescriptor> device,
audio_policy_dev_state_t state);
void updateMono(audio_io_handle_t output) {
AudioParameter param;
param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
mpClientInterface->setParameters(output, param.toString());
status_t installPatch(const char *caller,
audio_patch_handle_t *patchHandle,
AudioIODescriptorInterface *ioDescriptor,
const struct audio_patch *patch,
int delayMs);
status_t installPatch(const char *caller,
ssize_t index,
audio_patch_handle_t *patchHandle,
const struct audio_patch *patch,
int delayMs,
uid_t uid,
sp<AudioPatch> *patchDescPtr);
void cleanUpEffectsForIo(audio_io_handle_t io);