blob: 9af30dd34429080a019f1a92465ce70f984424ea [file] [log] [blame]
* 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.
#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;
// ----------------------------------------------------------------------------
// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
// Time in milliseconds during which we consider that music is still active after a music
// track was stopped - see computeVolume()
// 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.
// Time in milliseconds during witch some streams are muted while the audio path
// is switched
#define MUTE_TIME_MS 2000
// Default minimum length allowed for offloading a compressed track
// Can be overridden by the audio.offload.min.duration.secs property
// ----------------------------------------------------------------------------
// AudioPolicyManagerBase implements audio policy manager behavior common to all platforms.
// Each platform must implement an AudioPolicyManager class derived from AudioPolicyManagerBase
// and override methods for which the platform specific behavior differs from the implementation
// in AudioPolicyManagerBase. Even if no specific behavior is required, the AudioPolicyManager
// class must be implemented as well as the class factory function createAudioPolicyManager()
// and provided in a shared library
// ----------------------------------------------------------------------------
class AudioPolicyManagerBase: public AudioPolicyInterface
, public Thread
AudioPolicyManagerBase(AudioPolicyClientInterface *clientInterface);
virtual ~AudioPolicyManagerBase();
// AudioPolicyInterface
virtual status_t setDeviceConnectionState(audio_devices_t device,
AudioSystem::device_connection_state state,
const char *device_address);
virtual AudioSystem::device_connection_state getDeviceConnectionState(audio_devices_t device,
const char *device_address);
virtual void setPhoneState(int state);
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);
virtual status_t initCheck();
virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
AudioSystem::output_flags flags,
const audio_offload_info_t *offloadInfo);
virtual status_t startOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
audio_session_t session = AUDIO_SESSION_NONE);
virtual status_t stopOutput(audio_io_handle_t output,
AudioSystem::stream_type stream,
audio_session_t session = AUDIO_SESSION_NONE);
virtual void releaseOutput(audio_io_handle_t output);
virtual audio_io_handle_t getInput(int inputSource,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
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 closeAllInputs();
virtual void initStreamVolume(AudioSystem::stream_type stream,
int indexMin,
int indexMax);
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 audio_devices_t getDevicesForStream(AudioSystem::stream_type 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,
audio_session_t 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;
// 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(int 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);
enum routing_strategy {
// 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
int mIndex;
float mDBAttenuation;
// device categories used for volume curve management.
enum device_category {
class IOProfile;
class HwModule {
HwModule(const char *name);
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
IOProfile(HwModule *module);
bool isCompatibleProfile(audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
audio_channel_mask_t channelMask,
audio_output_flags_t flags) const;
void dump(int fd);
void log();
// 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 sSpeakerSonificationVolumeCurveDrc[AudioPolicyManagerBase::VOLCNT];
static const VolumeCurvePoint sDefaultSystemVolumeCurve[AudioPolicyManagerBase::VOLCNT];
static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[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[AudioSystem::NUM_STREAM_TYPES][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
AudioOutputDescriptor(const IOProfile *profile);
status_t dump(int fd);
audio_devices_t device() const;
void changeRefCount(AudioSystem::stream_type stream, int delta);
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 = 0) const;
bool isStreamActive(AudioSystem::stream_type stream,
uint32_t inPastMs = 0,
nsecs_t sysTime = 0) const;
bool isStrategyActive(routing_strategy strategy,
uint32_t inPastMs = 0,
nsecs_t sysTime = 0) const;
audio_io_handle_t mId; // output handle
uint32_t mSamplingRate; //
audio_format_t mFormat; //
audio_channel_mask_t mChannelMask; // output configuration
uint32_t mLatency; //
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()
uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
bool mForceRouting; // Next routing for this output will be forced as current device routed is null
// 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
AudioInputDescriptor(const IOProfile *profile);
status_t dump(int fd);
audio_io_handle_t mId; // input handle
uint32_t mSamplingRate; //
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)
const IOProfile *mProfile; // I/O profile this output derives from
// stream descriptor used for volume control
class 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
status_t dump(int fd);
int mIo; // io the effect is attached to
routing_strategy mStrategy; // routing strategy the effect is associated to
audio_session_t 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);
void addInput(audio_io_handle_t id, AudioInputDescriptor *inputDesc);
// 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 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);
// select input device corresponding to requested audio source
virtual audio_devices_t getDeviceForInputSource(int 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);
// 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, 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, 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(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 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(audio_devices_t device,
AudioSystem::device_connection_state state,
SortedVector<audio_io_handle_t>& outputs,
const String8 paramStr);
status_t checkInputsForDevice(audio_devices_t device,
AudioSystem::device_connection_state state,
SortedVector<audio_io_handle_t>& inputs,
const String8 paramStr);
// 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();
// 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 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 updateDevicesAndOutputs();
virtual uint32_t getMaxEffectsCpuLoad();
virtual uint32_t getMaxEffectsMemory();
virtual bool threadLoop();
void exit();
int testOutputIndex(audio_io_handle_t output);
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,
audio_format_t format,
audio_channel_mask_t channelMask);
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();
// Audio policy configuration file parsing (audio_policy.conf)
static uint32_t stringToEnum(const struct StringToEnum *table,
size_t size,
const char *name);
static bool stringToBool(const char *value);
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 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;
// list of input descriptors currently opened
DefaultKeyedVector<audio_io_handle_t, AudioInputDescriptor *> mInputs;
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
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 mUsbOutCardAndDevice; // USB audio ALSA card and device numbers:
// card=<card_number>;device=<><device_number>
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, 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)
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 <HwModule *> mHwModules;
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;
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);