Create a snapshot of the hardware_legacy
Create a snapshot of hardware_legacy for legacy HIDL, to avoid any build breakage due to new hardware_legacy for the new AIDL. Should be removed with the HIDL when we finish the switch
Bug: 205044134
Test: build and CtsWifiTest
Change-Id: Ib1068112f6c90f2a41b68e20027d959c95798120
diff --git a/wifi/1.6/default/hal_legacy/AudioPolicyManagerBase.h b/wifi/1.6/default/hal_legacy/AudioPolicyManagerBase.h
new file mode 100644
index 0000000..ccc0d32
--- /dev/null
+++ b/wifi/1.6/default/hal_legacy/AudioPolicyManagerBase.h
@@ -0,0 +1,567 @@
+/*
+ * 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 <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <hardware_legacy/AudioPolicyInterface.h>
+#include <stdint.h>
+#include <sys/types.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <utils/Timers.h>
+
+namespace android_audio_legacy {
+using android::DefaultKeyedVector;
+using android::KeyedVector;
+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,
+ 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);
+
+ 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)
+ 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 {
+ 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
+ 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();
+#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);
+};
+
+}; // namespace android_audio_legacy