/* * 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 <hardware_legacy/AudioPolicyInterface.h> namespace android_audio_legacy { using android::KeyedVector; using android::DefaultKeyedVector; using android::SortedVector; // ---------------------------------------------------------------------------- #define MAX_DEVICE_ADDRESS_LEN 20 // 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 // ---------------------------------------------------------------------------- // 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 libaudiopolicy.so. // ---------------------------------------------------------------------------- class AudioPolicyManagerBase: public AudioPolicyInterface #ifdef AUDIO_POLICY_TEST , public Thread #endif //AUDIO_POLICY_TEST { public: 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, int session = 0); virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0); 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, 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; // 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); 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_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, 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 { public: 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) }; // 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: 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 { 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: 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 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(); #ifdef AUDIO_POLICY_TEST virtual bool threadLoop(); void exit(); 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, 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; #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(AudioSystem::stream_type stream); static bool isVirtualInputDevice(audio_devices_t device); }; };