/*
* 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);
};
};