policy_hal: Initial change for the new libaudiopolicymanager
- Upgrade policy_hal to use new AudioPolicyManager
introduced by Google. The legacy AudioPolicyManagerBase
class is replaced by AudioPolicyManager.
- Customized AudioPolicyManager needs to implement everything
from /frameworks/av/service/audiopolicy/AudioPolicyManager
and add extended changes on top of it
- This change implements stock AOSP AudioPolicyManager with no
Additional changes.
Change-Id: I56f7c575e60c51876fc5eda59b2eaa29d4e77639
diff --git a/policy_hal/AudioPolicyManager.h b/policy_hal/AudioPolicyManager.h
index 9f2b473..0dd015a 100644
--- a/policy_hal/AudioPolicyManager.h
+++ b/policy_hal/AudioPolicyManager.h
@@ -1,5 +1,5 @@
/*
- * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Copyright (c) 2014, The Linux Foundation. All rights reserved.
* Not a contribution.
*
* Copyright (C) 2009 The Android Open Source Project
@@ -20,53 +20,470 @@
#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 <hardware_legacy/AudioPolicyManagerBase.h>
+#include <utils/SortedVector.h>
+#include "AudioPolicyInterface.h"
-namespace android_audio_legacy {
+namespace android {
// ----------------------------------------------------------------------------
-class AudioPolicyManager: public AudioPolicyManagerBase
+// 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
+
+// ----------------------------------------------------------------------------
+// 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)
- : AudioPolicyManagerBase(clientInterface) {
- mHdmiAudioDisabled = false;
- mHdmiAudioEvent = false; }
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+ virtual ~AudioPolicyManager();
- virtual ~AudioPolicyManager() {}
-
+ // AudioPolicyInterface
virtual status_t setDeviceConnectionState(audio_devices_t device,
- AudioSystem::device_connection_state state,
+ audio_policy_dev_state_t state,
const char *device_address);
- virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
- virtual audio_io_handle_t getInput(int inputSource,
+ 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,
- uint32_t format,
- uint32_t channels,
- AudioSystem::audio_in_acoustics acoustics);
- virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
- uint32_t samplingRate = 0,
- uint32_t format = AudioSystem::FORMAT_DEFAULT,
- uint32_t channels = 0,
- AudioSystem::output_flags flags =
- AudioSystem::OUTPUT_FLAG_INDIRECT,
- const audio_offload_info_t *offloadInfo = NULL);
+ 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_in_acoustics_t 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(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 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 void setPhoneState(int state);
+ 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);
- // true if given state represents a device in a telephony or VoIP call
- virtual bool isStateInCall(int state);
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 HwModule;
+
+ class AudioGain: public RefBase
+ {
+ public:
+ AudioGain();
+ virtual ~AudioGain() {}
+
+ void dump(int fd, int spaces, int index) const;
+
+ struct audio_gain mGain;
+ };
+
+ class AudioPort: public RefBase
+ {
+ public:
+ AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, HwModule *module) :
+ mName(name), mType(type), mRole(role), mModule(module) {}
+ virtual ~AudioPort() {}
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ 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);
+ void loadGains(cnode *root);
+
+ void dump(int fd, int spaces) const;
+
+ String8 mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ // 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
+ HwModule *mModule; // audio HW module exposing this I/O stream
+ };
+
+ 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) {}
+
+ audio_patch_handle_t mHandle;
+ struct audio_patch mPatch;
+ uid_t mUid;
+ audio_patch_handle_t mAfPatchHandle;
+ };
+
+ class DeviceDescriptor: public AudioPort
+ {
+ public:
+ DeviceDescriptor(const String8& name, audio_devices_t type, String8 address,
+ audio_channel_mask_t channelMask) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mDeviceType(type), mAddress(address),
+ mChannelMask(channelMask), mId(0) {}
+
+ DeviceDescriptor(String8 name, audio_devices_t type) :
+ AudioPort(name, AUDIO_PORT_TYPE_DEVICE,
+ audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
+ AUDIO_PORT_ROLE_SOURCE,
+ NULL),
+ mDeviceType(type), mAddress(""),
+ mChannelMask(0), mId(0) {}
+ virtual ~DeviceDescriptor() {}
+
+ bool equals(const sp<DeviceDescriptor>& other) const;
+ void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ 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_channel_mask_t mChannelMask;
+ 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;
+
+ 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, HwModule *module);
+ virtual ~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();
+
+ DeviceVector 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.
+ };
+
+ class HwModule {
+ 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 ...)
+ 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 media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[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:
+ 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 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;
+
+ void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ void toAudioPort(struct audio_port *port) const;
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // 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
+ 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];
+ AudioOutputDescriptor *mOutput1; // used by duplicated outputs: first output
+ 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:
+ AudioInputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // 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
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount; // number of AudioRecord clients using this output
+ audio_source_t mInputSource; // input source selected by application (mediarecorder.h)
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+
+ void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ 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:
+
+ 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, AudioOutputDescriptor *outputDesc);
+ void addInput(audio_io_handle_t input, AudioInputDescriptor *inputDesc);
+
// return the strategy corresponding to a given stream type
- static routing_strategy getStrategy(AudioSystem::stream_type stream);
+ 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...
@@ -80,32 +497,258 @@
// where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
// before updateDevicesAndOutputs() is called.
virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache = true);
+ 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);
+ 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(int inputSource);
+ 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);
+
+ // 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);
+ 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(int stream, int index, audio_io_handle_t output, audio_devices_t device, int delayMs = 0, bool force = false);
+ 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(audio_devices_t device,
+ 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);
+
+ // 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(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);
- static const char* HDMI_SPKR_STR;
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
- //parameter indicates of HDMI speakers disabled from the Qualcomm settings
- bool mHdmiAudioDisabled;
+ 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);
- //parameter indicates if HDMI plug in/out detected
- bool mHdmiAudioEvent;
+ // 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,
+ audio_output_flags_t flags);
+ sp<IOProfile> getInputProfile(audio_devices_t device,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask);
+ 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);
+
+ AudioOutputDescriptor *getOutputFromId(audio_port_handle_t id) const;
+ AudioInputDescriptor *getInputFromId(audio_port_handle_t id) const;
+ HwModule *getModuleForDevice(audio_devices_t device) const;
+ HwModule *getModuleFromName(const char *name) const;
+ //
+ // 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 audio_output_flags_t parseFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root, 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, AudioOutputDescriptor *> mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, AudioOutputDescriptor *> mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, 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, 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 <HwModule *> mHwModules;
+ volatile int32_t mNextUniqueId;
+ volatile int32_t mAudioPortGeneration;
+
+ DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
+
+#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:
- // Used for voip + voice concurrency usecase
- int mPrevPhoneState;
- static int mvoice_call_state;
-
+ 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);
+ uint32_t nextUniqueId();
+ uint32_t nextAudioPortGeneration();
+ uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // converts device address to string sent to audio HAL via setParameters
+ static String8 addressToParameter(audio_devices_t device, const String8 address);
};
+
};