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1045 lines
50 KiB
1045 lines
50 KiB
/*
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* Copyright (C) 2009 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#pragma once
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#include <atomic>
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#include <functional>
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#include <memory>
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#include <unordered_set>
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#include <stdint.h>
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#include <sys/types.h>
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#include <cutils/config_utils.h>
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#include <cutils/misc.h>
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#include <utils/Timers.h>
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#include <utils/Errors.h>
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#include <utils/KeyedVector.h>
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#include <utils/SortedVector.h>
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#include <media/AudioParameter.h>
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#include <media/AudioPolicy.h>
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#include <media/AudioProfile.h>
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#include <media/PatchBuilder.h>
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#include "AudioPolicyInterface.h"
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#include <AudioPolicyManagerObserver.h>
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#include <AudioPolicyConfig.h>
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#include <PolicyAudioPort.h>
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#include <AudioPatch.h>
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#include <DeviceDescriptor.h>
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#include <IOProfile.h>
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#include <HwModule.h>
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#include <AudioInputDescriptor.h>
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#include <AudioOutputDescriptor.h>
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#include <AudioPolicyMix.h>
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#include <EffectDescriptor.h>
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#include <SoundTriggerSession.h>
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#include "EngineLibrary.h"
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#include "TypeConverter.h"
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namespace android {
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using content::AttributionSourceState;
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// ----------------------------------------------------------------------------
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// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
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#define SONIFICATION_HEADSET_VOLUME_FACTOR_DB (-6)
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// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
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#define SONIFICATION_HEADSET_VOLUME_MIN_DB (-36)
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// Max volume difference on A2DP between playing media and STRATEGY_SONIFICATION streams: 12dB
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#define SONIFICATION_A2DP_MAX_MEDIA_DIFF_DB (12)
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// Time in milliseconds during which we consider that music is still active after a music
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// track was stopped - see computeVolume()
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#define SONIFICATION_HEADSET_MUSIC_DELAY 5000
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// Time in milliseconds during witch some streams are muted while the audio path
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// is switched
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#define MUTE_TIME_MS 2000
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// multiplication factor applied to output latency when calculating a safe mute delay when
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// invalidating tracks
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#define LATENCY_MUTE_FACTOR 4
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#define NUM_TEST_OUTPUTS 5
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#define NUM_VOL_CURVE_KNEES 2
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// Default minimum length allowed for offloading a compressed track
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// Can be overridden by the audio.offload.min.duration.secs property
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#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60
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// ----------------------------------------------------------------------------
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// AudioPolicyManager implements audio policy manager behavior common to all platforms.
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// ----------------------------------------------------------------------------
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class AudioPolicyManager : public AudioPolicyInterface, public AudioPolicyManagerObserver
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{
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public:
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explicit AudioPolicyManager(AudioPolicyClientInterface *clientInterface);
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virtual ~AudioPolicyManager();
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// AudioPolicyInterface
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virtual status_t setDeviceConnectionState(audio_devices_t device,
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audio_policy_dev_state_t state,
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const char *device_address,
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const char *device_name,
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audio_format_t encodedFormat);
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virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
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const char *device_address);
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virtual status_t handleDeviceConfigChange(audio_devices_t device,
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const char *device_address,
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const char *device_name,
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audio_format_t encodedFormat);
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virtual void setPhoneState(audio_mode_t state);
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virtual void setForceUse(audio_policy_force_use_t usage,
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audio_policy_forced_cfg_t config);
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virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
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virtual void setSystemProperty(const char* property, const char* value);
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virtual status_t initCheck();
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virtual audio_io_handle_t getOutput(audio_stream_type_t stream);
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status_t getOutputForAttr(const audio_attributes_t *attr,
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audio_io_handle_t *output,
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audio_session_t session,
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audio_stream_type_t *stream,
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const AttributionSourceState& attributionSource,
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const audio_config_t *config,
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audio_output_flags_t *flags,
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audio_port_handle_t *selectedDeviceId,
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audio_port_handle_t *portId,
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std::vector<audio_io_handle_t> *secondaryOutputs,
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output_type_t *outputType) override;
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virtual status_t startOutput(audio_port_handle_t portId);
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virtual status_t stopOutput(audio_port_handle_t portId);
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virtual bool releaseOutput(audio_port_handle_t portId);
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virtual status_t getInputForAttr(const audio_attributes_t *attr,
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audio_io_handle_t *input,
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audio_unique_id_t riid,
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audio_session_t session,
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const AttributionSourceState& attributionSource,
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const audio_config_base_t *config,
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audio_input_flags_t flags,
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audio_port_handle_t *selectedDeviceId,
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input_type_t *inputType,
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audio_port_handle_t *portId);
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// indicates to the audio policy manager that the input starts being used.
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virtual status_t startInput(audio_port_handle_t portId);
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// indicates to the audio policy manager that the input stops being used.
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virtual status_t stopInput(audio_port_handle_t portId);
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virtual void releaseInput(audio_port_handle_t portId);
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virtual void checkCloseInputs();
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/**
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* @brief initStreamVolume: even if the engine volume files provides min and max, keep this
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* api for compatibility reason.
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* AudioServer will get the min and max and may overwrite them if:
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* -using property (highest priority)
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* -not defined (-1 by convention), case when still using apm volume tables XML files
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* @param stream to be considered
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* @param indexMin to set
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* @param indexMax to set
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*/
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virtual void initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax);
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virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
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int index,
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audio_devices_t device);
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virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
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int *index,
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audio_devices_t device);
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virtual status_t setVolumeIndexForAttributes(const audio_attributes_t &attr,
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int index,
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audio_devices_t device);
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virtual status_t getVolumeIndexForAttributes(const audio_attributes_t &attr,
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int &index,
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audio_devices_t device);
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virtual status_t getMaxVolumeIndexForAttributes(const audio_attributes_t &attr, int &index);
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virtual status_t getMinVolumeIndexForAttributes(const audio_attributes_t &attr, int &index);
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status_t setVolumeCurveIndex(int index,
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audio_devices_t device,
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IVolumeCurves &volumeCurves);
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status_t getVolumeIndex(const IVolumeCurves &curves, int &index,
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const DeviceTypeSet& deviceTypes) const;
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// return the strategy corresponding to a given stream type
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virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream)
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{
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return streamToStrategy(stream);
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}
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product_strategy_t streamToStrategy(audio_stream_type_t stream) const
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{
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auto attributes = mEngine->getAttributesForStreamType(stream);
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return mEngine->getProductStrategyForAttributes(attributes);
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}
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// return the enabled output devices for the given stream type
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virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
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virtual status_t getDevicesForAttributes(
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const audio_attributes_t &attributes,
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AudioDeviceTypeAddrVector *devices);
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virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc = NULL);
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virtual status_t registerEffect(const effect_descriptor_t *desc,
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audio_io_handle_t io,
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product_strategy_t strategy,
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int session,
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int id);
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virtual status_t unregisterEffect(int id);
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virtual status_t setEffectEnabled(int id, bool enabled);
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status_t moveEffectsToIo(const std::vector<int>& ids, audio_io_handle_t io) override;
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virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
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// return whether a stream is playing remotely, override to change the definition of
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// local/remote playback, used for instance by notification manager to not make
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// media players lose audio focus when not playing locally
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// For the base implementation, "remotely" means playing during screen mirroring which
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// uses an output for playback with a non-empty, non "0" address.
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virtual bool isStreamActiveRemotely(audio_stream_type_t stream,
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uint32_t inPastMs = 0) const;
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virtual bool isSourceActive(audio_source_t source) const;
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// helpers for dump(int fd)
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void dumpManualSurroundFormats(String8 *dst) const;
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void dump(String8 *dst) const;
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status_t dump(int fd) override;
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status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t capturePolicy) override;
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virtual audio_offload_mode_t getOffloadSupport(const audio_offload_info_t& offloadInfo);
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virtual bool isDirectOutputSupported(const audio_config_base_t& config,
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const audio_attributes_t& attributes);
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virtual status_t listAudioPorts(audio_port_role_t role,
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audio_port_type_t type,
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unsigned int *num_ports,
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struct audio_port_v7 *ports,
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unsigned int *generation);
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virtual status_t getAudioPort(struct audio_port_v7 *port);
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virtual status_t createAudioPatch(const struct audio_patch *patch,
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audio_patch_handle_t *handle,
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uid_t uid) {
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return createAudioPatchInternal(patch, handle, uid);
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}
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virtual status_t releaseAudioPatch(audio_patch_handle_t handle,
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uid_t uid);
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virtual status_t listAudioPatches(unsigned int *num_patches,
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struct audio_patch *patches,
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unsigned int *generation);
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virtual status_t setAudioPortConfig(const struct audio_port_config *config);
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virtual void releaseResourcesForUid(uid_t uid);
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virtual status_t acquireSoundTriggerSession(audio_session_t *session,
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audio_io_handle_t *ioHandle,
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audio_devices_t *device);
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virtual status_t releaseSoundTriggerSession(audio_session_t session)
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{
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return mSoundTriggerSessions.releaseSession(session);
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}
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virtual status_t registerPolicyMixes(const Vector<AudioMix>& mixes);
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virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
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virtual status_t setUidDeviceAffinities(uid_t uid,
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const AudioDeviceTypeAddrVector& devices);
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virtual status_t removeUidDeviceAffinities(uid_t uid);
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virtual status_t setUserIdDeviceAffinities(int userId,
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const AudioDeviceTypeAddrVector& devices);
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virtual status_t removeUserIdDeviceAffinities(int userId);
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virtual status_t setDevicesRoleForStrategy(product_strategy_t strategy,
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device_role_t role,
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const AudioDeviceTypeAddrVector &devices);
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virtual status_t removeDevicesRoleForStrategy(product_strategy_t strategy,
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device_role_t role);
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virtual status_t getDevicesForRoleAndStrategy(product_strategy_t strategy,
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device_role_t role,
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AudioDeviceTypeAddrVector &devices);
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virtual status_t setDevicesRoleForCapturePreset(audio_source_t audioSource,
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device_role_t role,
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const AudioDeviceTypeAddrVector &devices);
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virtual status_t addDevicesRoleForCapturePreset(audio_source_t audioSource,
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device_role_t role,
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const AudioDeviceTypeAddrVector &devices);
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virtual status_t removeDevicesRoleForCapturePreset(
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audio_source_t audioSource, device_role_t role,
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const AudioDeviceTypeAddrVector& devices);
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virtual status_t clearDevicesRoleForCapturePreset(audio_source_t audioSource,
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device_role_t role);
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virtual status_t getDevicesForRoleAndCapturePreset(audio_source_t audioSource,
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device_role_t role,
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AudioDeviceTypeAddrVector &devices);
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virtual status_t startAudioSource(const struct audio_port_config *source,
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const audio_attributes_t *attributes,
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audio_port_handle_t *portId,
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uid_t uid);
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virtual status_t stopAudioSource(audio_port_handle_t portId);
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virtual status_t setMasterMono(bool mono);
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virtual status_t getMasterMono(bool *mono);
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virtual float getStreamVolumeDB(
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audio_stream_type_t stream, int index, audio_devices_t device);
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virtual status_t getSurroundFormats(unsigned int *numSurroundFormats,
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audio_format_t *surroundFormats,
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bool *surroundFormatsEnabled);
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virtual status_t getReportedSurroundFormats(unsigned int *numSurroundFormats,
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audio_format_t *surroundFormats);
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virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
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virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
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std::vector<audio_format_t> *formats);
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virtual void setAppState(audio_port_handle_t portId, app_state_t state);
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virtual bool isHapticPlaybackSupported();
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virtual status_t listAudioProductStrategies(AudioProductStrategyVector &strategies)
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{
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return mEngine->listAudioProductStrategies(strategies);
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}
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virtual status_t getProductStrategyFromAudioAttributes(
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const AudioAttributes &aa, product_strategy_t &productStrategy,
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bool fallbackOnDefault)
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{
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productStrategy = mEngine->getProductStrategyForAttributes(
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aa.getAttributes(), fallbackOnDefault);
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return (fallbackOnDefault && productStrategy == PRODUCT_STRATEGY_NONE) ?
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BAD_VALUE : NO_ERROR;
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}
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virtual status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups)
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{
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return mEngine->listAudioVolumeGroups(groups);
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}
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virtual status_t getVolumeGroupFromAudioAttributes(
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const AudioAttributes &aa, volume_group_t &volumeGroup, bool fallbackOnDefault)
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{
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volumeGroup = mEngine->getVolumeGroupForAttributes(
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aa.getAttributes(), fallbackOnDefault);
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return (fallbackOnDefault && volumeGroup == VOLUME_GROUP_NONE) ?
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BAD_VALUE : NO_ERROR;
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}
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bool isCallScreenModeSupported() override;
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void onNewAudioModulesAvailable() override;
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status_t initialize();
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protected:
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// A constructor that allows more fine-grained control over initialization process,
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// used in automatic tests.
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AudioPolicyManager(AudioPolicyClientInterface *clientInterface, bool forTesting);
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// These methods should be used when finer control over APM initialization
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// is needed, e.g. in tests. Must be used in conjunction with the constructor
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// that only performs fields initialization. The public constructor comprises
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// these steps in the following sequence:
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// - field initializing constructor;
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// - loadConfig;
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// - initialize.
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AudioPolicyConfig& getConfig() { return mConfig; }
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void loadConfig();
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// From AudioPolicyManagerObserver
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virtual const AudioPatchCollection &getAudioPatches() const
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{
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return mAudioPatches;
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}
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virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const
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{
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return mSoundTriggerSessions;
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}
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virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const
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{
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return mPolicyMixes;
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}
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virtual const SwAudioOutputCollection &getOutputs() const
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{
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return mOutputs;
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}
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virtual const AudioInputCollection &getInputs() const
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{
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return mInputs;
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}
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virtual const DeviceVector getAvailableOutputDevices() const
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{
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return mAvailableOutputDevices.filterForEngine();
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}
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virtual const DeviceVector getAvailableInputDevices() const
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{
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// legacy and non-legacy remote-submix are managed by the engine, do not filter
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return mAvailableInputDevices;
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}
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virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const
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{
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return mDefaultOutputDevice;
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}
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std::vector<volume_group_t> getVolumeGroups() const
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{
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return mEngine->getVolumeGroups();
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}
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VolumeSource toVolumeSource(volume_group_t volumeGroup) const
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{
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return static_cast<VolumeSource>(volumeGroup);
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}
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VolumeSource toVolumeSource(const audio_attributes_t &attributes) const
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{
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return toVolumeSource(mEngine->getVolumeGroupForAttributes(attributes));
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}
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VolumeSource toVolumeSource(audio_stream_type_t stream) const
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{
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return toVolumeSource(mEngine->getVolumeGroupForStreamType(stream));
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}
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IVolumeCurves &getVolumeCurves(VolumeSource volumeSource)
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{
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auto *curves = mEngine->getVolumeCurvesForVolumeGroup(
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static_cast<volume_group_t>(volumeSource));
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ALOG_ASSERT(curves != nullptr, "No curves for volume source %d", volumeSource);
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return *curves;
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}
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IVolumeCurves &getVolumeCurves(const audio_attributes_t &attr)
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{
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auto *curves = mEngine->getVolumeCurvesForAttributes(attr);
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ALOG_ASSERT(curves != nullptr, "No curves for attributes %s", toString(attr).c_str());
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return *curves;
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}
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IVolumeCurves &getVolumeCurves(audio_stream_type_t stream)
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{
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auto *curves = mEngine->getVolumeCurvesForStreamType(stream);
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ALOG_ASSERT(curves != nullptr, "No curves for stream %s", toString(stream).c_str());
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return *curves;
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}
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void addOutput(audio_io_handle_t output, const sp<SwAudioOutputDescriptor>& outputDesc);
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void removeOutput(audio_io_handle_t output);
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void addInput(audio_io_handle_t input, const sp<AudioInputDescriptor>& inputDesc);
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// change the route of the specified output. Returns the number of ms we have slept to
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// allow new routing to take effect in certain cases.
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uint32_t setOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
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const DeviceVector &device,
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bool force = false,
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int delayMs = 0,
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audio_patch_handle_t *patchHandle = NULL,
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bool requiresMuteCheck = true);
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status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc,
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int delayMs = 0,
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audio_patch_handle_t *patchHandle = NULL);
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status_t setInputDevice(audio_io_handle_t input,
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const sp<DeviceDescriptor> &device,
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bool force = false,
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audio_patch_handle_t *patchHandle = NULL);
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status_t resetInputDevice(audio_io_handle_t input,
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audio_patch_handle_t *patchHandle = NULL);
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// compute the actual volume for a given stream according to the requested index and a particular
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// device
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virtual float computeVolume(IVolumeCurves &curves,
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VolumeSource volumeSource,
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int index,
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const DeviceTypeSet& deviceTypes);
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// rescale volume index from srcStream within range of dstStream
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int rescaleVolumeIndex(int srcIndex,
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VolumeSource fromVolumeSource,
|
|
VolumeSource toVolumeSource);
|
|
// check that volume change is permitted, compute and send new volume to audio hardware
|
|
virtual status_t checkAndSetVolume(IVolumeCurves &curves,
|
|
VolumeSource volumeSource, int index,
|
|
const sp<AudioOutputDescriptor>& outputDesc,
|
|
DeviceTypeSet deviceTypes,
|
|
int delayMs = 0, bool force = false);
|
|
|
|
// apply all stream volumes to the specified output and device
|
|
void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc,
|
|
const DeviceTypeSet& deviceTypes,
|
|
int delayMs = 0, bool force = false);
|
|
|
|
/**
|
|
* @brief setStrategyMute Mute or unmute all active clients on the considered output
|
|
* following the given strategy.
|
|
* @param strategy to be considered
|
|
* @param on true for mute, false for unmute
|
|
* @param outputDesc to be considered
|
|
* @param delayMs
|
|
* @param device
|
|
*/
|
|
void setStrategyMute(product_strategy_t strategy,
|
|
bool on,
|
|
const sp<AudioOutputDescriptor>& outputDesc,
|
|
int delayMs = 0,
|
|
DeviceTypeSet deviceTypes = DeviceTypeSet());
|
|
|
|
/**
|
|
* @brief setVolumeSourceMute Mute or unmute the volume source on the specified output
|
|
* @param volumeSource to be muted/unmute (may host legacy streams or by extension set of
|
|
* audio attributes)
|
|
* @param on true to mute, false to umute
|
|
* @param outputDesc on which the client following the volume group shall be muted/umuted
|
|
* @param delayMs
|
|
* @param device
|
|
*/
|
|
void setVolumeSourceMute(VolumeSource volumeSource,
|
|
bool on,
|
|
const sp<AudioOutputDescriptor>& outputDesc,
|
|
int delayMs = 0,
|
|
DeviceTypeSet deviceTypes = DeviceTypeSet());
|
|
|
|
audio_mode_t getPhoneState();
|
|
|
|
// 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);
|
|
// true if playback to call TX or capture from call RX is possible
|
|
bool isCallAudioAccessible();
|
|
|
|
// when a device is connected, checks if an open output can be routed
|
|
// to this device. If none is open, tries to open one of the available outputs.
|
|
// Returns an output suitable to this device or 0.
|
|
// when a device is disconnected, checks if an output is not used any more and
|
|
// returns its handle if any.
|
|
// transfers the audio tracks and effects from one output thread to another accordingly.
|
|
status_t checkOutputsForDevice(const sp<DeviceDescriptor>& device,
|
|
audio_policy_dev_state_t state,
|
|
SortedVector<audio_io_handle_t>& outputs);
|
|
|
|
status_t checkInputsForDevice(const sp<DeviceDescriptor>& device,
|
|
audio_policy_dev_state_t state);
|
|
|
|
// close an output and its companion duplicating output.
|
|
void closeOutput(audio_io_handle_t output);
|
|
|
|
// close an input.
|
|
void closeInput(audio_io_handle_t input);
|
|
|
|
// runs all the checks required for accommodating changes in devices and outputs
|
|
// if 'onOutputsChecked' callback is provided, it is executed after the outputs
|
|
// check via 'checkOutputForAllStrategies'. If the callback returns 'true',
|
|
// A2DP suspend status is rechecked.
|
|
void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked = nullptr);
|
|
|
|
/**
|
|
* @brief updates routing for all outputs (including call if call in progress).
|
|
* @param delayMs delay for unmuting if required
|
|
*/
|
|
void updateCallAndOutputRouting(bool forceVolumeReeval = true, uint32_t delayMs = 0);
|
|
|
|
bool isCallRxAudioSource(const sp<SourceClientDescriptor> &source) {
|
|
return mCallRxSourceClientPort != AUDIO_PORT_HANDLE_NONE
|
|
&& source == mAudioSources.valueFor(mCallRxSourceClientPort);
|
|
}
|
|
|
|
void connectTelephonyRxAudioSource();
|
|
|
|
void disconnectTelephonyRxAudioSource();
|
|
|
|
/**
|
|
* @brief updates routing for all inputs.
|
|
*/
|
|
void updateInputRouting();
|
|
|
|
/**
|
|
* @brief checkOutputForAttributes checks and if necessary changes outputs used for the
|
|
* given audio attributes.
|
|
* must be called every time a condition that affects the output choice for a given
|
|
* attributes changes: connected device, phone state, force use...
|
|
* Must be called before updateDevicesAndOutputs()
|
|
* @param attr to be considered
|
|
*/
|
|
void checkOutputForAttributes(const audio_attributes_t &attr);
|
|
|
|
/**
|
|
* @brief checkAudioSourceForAttributes checks if any AudioSource following the same routing
|
|
* as the given audio attributes is not routed and try to connect it.
|
|
* It must be called once checkOutputForAttributes has been called for orphans AudioSource,
|
|
* aka AudioSource not attached to any Audio Output (e.g. AudioSource connected to direct
|
|
* Output which has been disconnected (and output closed) due to sink device unavailable).
|
|
* @param attr to be considered
|
|
*/
|
|
void checkAudioSourceForAttributes(const audio_attributes_t &attr);
|
|
|
|
bool followsSameRouting(const audio_attributes_t &lAttr,
|
|
const audio_attributes_t &rAttr) const;
|
|
|
|
/**
|
|
* @brief checkOutputForAllStrategies Same as @see checkOutputForAttributes()
|
|
* but for a all product strategies in order of priority
|
|
*/
|
|
void checkOutputForAllStrategies();
|
|
|
|
// Same as checkOutputForStrategy but for secondary outputs. Make sure if a secondary
|
|
// output condition changes, the track is properly rerouted
|
|
void checkSecondaryOutputs();
|
|
|
|
// manages A2DP output suspend/restore according to phone state and BT SCO usage
|
|
void checkA2dpSuspend();
|
|
|
|
// 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
|
|
DeviceVector getNewOutputDevices(const sp<SwAudioOutputDescriptor>& outputDesc,
|
|
bool fromCache);
|
|
|
|
/**
|
|
* @brief updateDevicesAndOutputs: updates cache of devices of the engine
|
|
* must be called every time a condition that affects the device choice is changed:
|
|
* connected device, phone state, force use...
|
|
* cached values are used by getOutputDevicesForStream()/getDevicesForAttributes if
|
|
* parameter fromCache is true.
|
|
* Must be called after checkOutputForAllStrategies()
|
|
*/
|
|
void updateDevicesAndOutputs();
|
|
|
|
// selects the most appropriate device on input for current state
|
|
sp<DeviceDescriptor> getNewInputDevice(const sp<AudioInputDescriptor>& inputDesc);
|
|
|
|
virtual uint32_t getMaxEffectsCpuLoad()
|
|
{
|
|
return mEffects.getMaxEffectsCpuLoad();
|
|
}
|
|
|
|
virtual uint32_t getMaxEffectsMemory()
|
|
{
|
|
return mEffects.getMaxEffectsMemory();
|
|
}
|
|
|
|
SortedVector<audio_io_handle_t> getOutputsForDevices(
|
|
const DeviceVector &devices, const SwAudioOutputCollection& openOutputs);
|
|
|
|
/**
|
|
* @brief checkDeviceMuteStrategies 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
|
|
* @param outputDesc
|
|
* @param prevDevice
|
|
* @param delayMs
|
|
* @return the number of ms waited
|
|
*/
|
|
virtual uint32_t checkDeviceMuteStrategies(const sp<AudioOutputDescriptor>& outputDesc,
|
|
const DeviceVector &prevDevices,
|
|
uint32_t delayMs);
|
|
|
|
audio_io_handle_t selectOutput(const SortedVector<audio_io_handle_t>& outputs,
|
|
audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
|
|
audio_format_t format = AUDIO_FORMAT_INVALID,
|
|
audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE,
|
|
uint32_t samplingRate = 0,
|
|
audio_session_t sessionId = AUDIO_SESSION_NONE);
|
|
// samplingRate, format, channelMask are in/out and so may be modified
|
|
sp<IOProfile> getInputProfile(const sp<DeviceDescriptor> & device,
|
|
uint32_t& samplingRate,
|
|
audio_format_t& format,
|
|
audio_channel_mask_t& channelMask,
|
|
audio_input_flags_t flags);
|
|
/**
|
|
* @brief getProfileForOutput
|
|
* @param devices vector of descriptors, may be empty if ignoring the device is required
|
|
* @param samplingRate
|
|
* @param format
|
|
* @param channelMask
|
|
* @param flags
|
|
* @param directOnly
|
|
* @return IOProfile to be used if found, nullptr otherwise
|
|
*/
|
|
sp<IOProfile> getProfileForOutput(const DeviceVector &devices,
|
|
uint32_t samplingRate,
|
|
audio_format_t format,
|
|
audio_channel_mask_t channelMask,
|
|
audio_output_flags_t flags,
|
|
bool directOnly);
|
|
|
|
audio_io_handle_t selectOutputForMusicEffects();
|
|
|
|
virtual status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch)
|
|
{
|
|
return mAudioPatches.addAudioPatch(handle, patch);
|
|
}
|
|
virtual status_t removeAudioPatch(audio_patch_handle_t handle)
|
|
{
|
|
return mAudioPatches.removeAudioPatch(handle);
|
|
}
|
|
|
|
bool isPrimaryModule(const sp<HwModule> &module) const
|
|
{
|
|
if (module == 0 || !hasPrimaryOutput()) {
|
|
return false;
|
|
}
|
|
return module->getHandle() == mPrimaryOutput->getModuleHandle();
|
|
}
|
|
DeviceVector availablePrimaryOutputDevices() const
|
|
{
|
|
if (!hasPrimaryOutput()) {
|
|
return DeviceVector();
|
|
}
|
|
return mAvailableOutputDevices.filter(mPrimaryOutput->supportedDevices());
|
|
}
|
|
DeviceVector availablePrimaryModuleInputDevices() const
|
|
{
|
|
if (!hasPrimaryOutput()) {
|
|
return DeviceVector();
|
|
}
|
|
return mAvailableInputDevices.getDevicesFromHwModule(
|
|
mPrimaryOutput->getModuleHandle());
|
|
}
|
|
/**
|
|
* @brief getFirstDeviceId of the Device Vector
|
|
* @return if the collection is not empty, it returns the first device Id,
|
|
* otherwise AUDIO_PORT_HANDLE_NONE
|
|
*/
|
|
audio_port_handle_t getFirstDeviceId(const DeviceVector &devices) const
|
|
{
|
|
return (devices.size() > 0) ? devices.itemAt(0)->getId() : AUDIO_PORT_HANDLE_NONE;
|
|
}
|
|
String8 getFirstDeviceAddress(const DeviceVector &devices) const
|
|
{
|
|
return (devices.size() > 0) ?
|
|
String8(devices.itemAt(0)->address().c_str()) : String8("");
|
|
}
|
|
|
|
status_t updateCallRouting(
|
|
bool fromCache, uint32_t delayMs = 0, uint32_t *waitMs = nullptr);
|
|
status_t updateCallRoutingInternal(
|
|
const DeviceVector &rxDevices, uint32_t delayMs, uint32_t *waitMs);
|
|
sp<AudioPatch> createTelephonyPatch(bool isRx, const sp<DeviceDescriptor> &device,
|
|
uint32_t delayMs);
|
|
/**
|
|
* @brief selectBestRxSinkDevicesForCall: if the primary module host both Telephony Rx/Tx
|
|
* devices, and it declares also supporting a HW bridge between the Telephony Rx and the
|
|
* given sink device for Voice Call audio attributes, select this device in prio.
|
|
* Otherwise, getNewOutputDevices() is called on the primary output to select sink device.
|
|
* @param fromCache true to prevent engine reconsidering all product strategies and retrieve
|
|
* from engine cache.
|
|
* @return vector of devices, empty if none is found.
|
|
*/
|
|
DeviceVector selectBestRxSinkDevicesForCall(bool fromCache);
|
|
bool isDeviceOfModule(const sp<DeviceDescriptor>& devDesc, const char *moduleId) const;
|
|
|
|
status_t startSource(const sp<SwAudioOutputDescriptor>& outputDesc,
|
|
const sp<TrackClientDescriptor>& client,
|
|
uint32_t *delayMs);
|
|
status_t stopSource(const sp<SwAudioOutputDescriptor>& outputDesc,
|
|
const sp<TrackClientDescriptor>& client);
|
|
|
|
void clearAudioPatches(uid_t uid);
|
|
void clearSessionRoutes(uid_t uid);
|
|
|
|
/**
|
|
* @brief checkStrategyRoute: when an output is beeing rerouted, reconsider each output
|
|
* that may host a strategy playing on the considered output.
|
|
* @param ps product strategy that initiated the rerouting
|
|
* @param ouptutToSkip output that initiated the rerouting
|
|
*/
|
|
void checkStrategyRoute(product_strategy_t ps, audio_io_handle_t ouptutToSkip);
|
|
|
|
status_t hasPrimaryOutput() const { return mPrimaryOutput != 0; }
|
|
|
|
status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
|
|
status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
|
|
|
|
sp<SourceClientDescriptor> getSourceForAttributesOnOutput(audio_io_handle_t output,
|
|
const audio_attributes_t &attr);
|
|
void clearAudioSourcesForOutput(audio_io_handle_t output);
|
|
|
|
void cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc);
|
|
|
|
void clearAudioSources(uid_t uid);
|
|
|
|
static bool streamsMatchForvolume(audio_stream_type_t stream1,
|
|
audio_stream_type_t stream2);
|
|
|
|
void closeActiveClients(const sp<AudioInputDescriptor>& input);
|
|
void closeClient(audio_port_handle_t portId);
|
|
|
|
const uid_t mUidCached; // AID_AUDIOSERVER
|
|
AudioPolicyClientInterface *mpClientInterface; // audio policy client interface
|
|
sp<SwAudioOutputDescriptor> mPrimaryOutput; // primary output descriptor
|
|
// list of descriptors for outputs currently opened
|
|
|
|
SwAudioOutputCollection mOutputs;
|
|
// copy of mOutputs before setDeviceConnectionState() opens new outputs
|
|
// reset to mOutputs when updateDevicesAndOutputs() is called.
|
|
SwAudioOutputCollection mPreviousOutputs;
|
|
AudioInputCollection mInputs; // list of input descriptors
|
|
|
|
DeviceVector mOutputDevicesAll; // all output devices from the config
|
|
DeviceVector mInputDevicesAll; // all input devices from the config
|
|
DeviceVector mAvailableOutputDevices; // all available output devices
|
|
DeviceVector mAvailableInputDevices; // all available input devices
|
|
|
|
bool mLimitRingtoneVolume; // limit ringtone volume to music volume if headset connected
|
|
|
|
float mLastVoiceVolume; // last voice volume value sent to audio HAL
|
|
bool mA2dpSuspended; // true if A2DP output is suspended
|
|
|
|
EffectDescriptorCollection mEffects; // list of registered audio effects
|
|
sp<DeviceDescriptor> mDefaultOutputDevice; // output device selected by default at boot time
|
|
HwModuleCollection mHwModules; // contains modules that have been loaded successfully
|
|
HwModuleCollection mHwModulesAll; // contains all modules declared in the config
|
|
|
|
AudioPolicyConfig mConfig;
|
|
|
|
std::atomic<uint32_t> mAudioPortGeneration;
|
|
|
|
AudioPatchCollection mAudioPatches;
|
|
|
|
SoundTriggerSessionCollection mSoundTriggerSessions;
|
|
|
|
sp<AudioPatch> mCallTxPatch;
|
|
|
|
HwAudioOutputCollection mHwOutputs;
|
|
SourceClientCollection mAudioSources;
|
|
|
|
// for supporting "beacon" streams, i.e. streams that only play on speaker, and never
|
|
// when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
|
|
enum {
|
|
STARTING_OUTPUT,
|
|
STARTING_BEACON,
|
|
STOPPING_OUTPUT,
|
|
STOPPING_BEACON
|
|
};
|
|
uint32_t mBeaconMuteRefCount; // ref count for stream that would mute beacon
|
|
uint32_t mBeaconPlayingRefCount;// ref count for the playing beacon streams
|
|
bool mBeaconMuted; // has STREAM_TTS been muted
|
|
bool mTtsOutputAvailable; // true if a dedicated output for TTS stream is available
|
|
|
|
bool mMasterMono; // true if we wish to force all outputs to mono
|
|
AudioPolicyMixCollection mPolicyMixes; // list of registered mixes
|
|
audio_io_handle_t mMusicEffectOutput; // output selected for music effects
|
|
|
|
uint32_t nextAudioPortGeneration();
|
|
|
|
// Audio Policy Engine Interface.
|
|
EngineInstance mEngine;
|
|
|
|
// Surround formats that are enabled manually. Taken into account when
|
|
// "encoded surround" is forced into "manual" mode.
|
|
std::unordered_set<audio_format_t> mManualSurroundFormats;
|
|
|
|
std::unordered_map<uid_t, audio_flags_mask_t> mAllowedCapturePolicies;
|
|
|
|
// The map of device descriptor and formats reported by the device.
|
|
std::map<wp<DeviceDescriptor>, FormatVector> mReportedFormatsMap;
|
|
|
|
// Cached product strategy ID corresponding to legacy strategy STRATEGY_PHONE
|
|
product_strategy_t mCommunnicationStrategy;
|
|
|
|
// The port handle of the hardware audio source created internally for the Call RX audio
|
|
// end point.
|
|
audio_port_handle_t mCallRxSourceClientPort = AUDIO_PORT_HANDLE_NONE;
|
|
|
|
// Support for Multi-Stream Decoder (MSD) module
|
|
sp<DeviceDescriptor> getMsdAudioInDevice() const;
|
|
DeviceVector getMsdAudioOutDevices() const;
|
|
const AudioPatchCollection getMsdOutputPatches() const;
|
|
status_t getMsdProfiles(bool hwAvSync,
|
|
const InputProfileCollection &inputProfiles,
|
|
const OutputProfileCollection &outputProfiles,
|
|
const sp<DeviceDescriptor> &sourceDevice,
|
|
const sp<DeviceDescriptor> &sinkDevice,
|
|
AudioProfileVector &sourceProfiles,
|
|
AudioProfileVector &sinkProfiles) const;
|
|
status_t getBestMsdConfig(bool hwAvSync,
|
|
const AudioProfileVector &sourceProfiles,
|
|
const AudioProfileVector &sinkProfiles,
|
|
audio_port_config *sourceConfig,
|
|
audio_port_config *sinkConfig) const;
|
|
PatchBuilder buildMsdPatch(bool msdIsSource, const sp<DeviceDescriptor> &device) const;
|
|
status_t setMsdOutputPatches(const DeviceVector *outputDevices = nullptr);
|
|
void releaseMsdOutputPatches(const DeviceVector& devices);
|
|
private:
|
|
void onNewAudioModulesAvailableInt(DeviceVector *newDevices);
|
|
|
|
// Add or remove AC3 DTS encodings based on user preferences.
|
|
void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
|
|
void modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr);
|
|
|
|
// If any, resolve any "dynamic" fields of an Audio Profiles collection
|
|
void updateAudioProfiles(const sp<DeviceDescriptor>& devDesc, audio_io_handle_t ioHandle,
|
|
AudioProfileVector &profiles);
|
|
|
|
// Notify the policy client of any change of device state with AUDIO_IO_HANDLE_NONE,
|
|
// so that the client interprets it as global to audio hardware interfaces.
|
|
// It can give a chance to HAL implementer to retrieve dynamic capabilities associated
|
|
// to this device for example.
|
|
// TODO avoid opening stream to retrieve capabilities of a profile.
|
|
void broadcastDeviceConnectionState(const sp<DeviceDescriptor> &device,
|
|
audio_policy_dev_state_t state);
|
|
|
|
// updates device caching and output for streams that can influence the
|
|
// routing of notifications
|
|
void handleNotificationRoutingForStream(audio_stream_type_t stream);
|
|
uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
|
|
// internal method, get audio_attributes_t from either a source audio_attributes_t
|
|
// or audio_stream_type_t, respectively.
|
|
status_t getAudioAttributes(audio_attributes_t *dstAttr,
|
|
const audio_attributes_t *srcAttr,
|
|
audio_stream_type_t srcStream);
|
|
// internal method, called by getOutputForAttr() and connectAudioSource.
|
|
status_t getOutputForAttrInt(audio_attributes_t *resultAttr,
|
|
audio_io_handle_t *output,
|
|
audio_session_t session,
|
|
const audio_attributes_t *attr,
|
|
audio_stream_type_t *stream,
|
|
uid_t uid,
|
|
const audio_config_t *config,
|
|
audio_output_flags_t *flags,
|
|
audio_port_handle_t *selectedDeviceId,
|
|
bool *isRequestedDeviceForExclusiveUse,
|
|
std::vector<sp<AudioPolicyMix>> *secondaryMixes,
|
|
output_type_t *outputType);
|
|
// internal method to return the output handle for the given device and format
|
|
audio_io_handle_t getOutputForDevices(
|
|
const DeviceVector &devices,
|
|
audio_session_t session,
|
|
audio_stream_type_t stream,
|
|
const audio_config_t *config,
|
|
audio_output_flags_t *flags,
|
|
bool forceMutingHaptic = false);
|
|
|
|
// Internal method checking if a direct output can be opened matching the requested
|
|
// attributes, flags, config and devices.
|
|
// If NAME_NOT_FOUND is returned, an attempt can be made to open a mixed output.
|
|
status_t openDirectOutput(
|
|
audio_stream_type_t stream,
|
|
audio_session_t session,
|
|
const audio_config_t *config,
|
|
audio_output_flags_t flags,
|
|
const DeviceVector &devices,
|
|
audio_io_handle_t *output);
|
|
/**
|
|
* @brief getInputForDevice selects an input handle for a given input device and
|
|
* requester context
|
|
* @param device to be used by requester, selected by policy mix rules or engine
|
|
* @param session requester session id
|
|
* @param uid requester uid
|
|
* @param attributes requester audio attributes (e.g. input source and tags matter)
|
|
* @param config requester audio configuration (e.g. sample rate, format, channel mask).
|
|
* @param flags requester input flags
|
|
* @param policyMix may be null, policy rules to be followed by the requester
|
|
* @return input io handle aka unique input identifier selected for this device.
|
|
*/
|
|
audio_io_handle_t getInputForDevice(const sp<DeviceDescriptor> &device,
|
|
audio_session_t session,
|
|
const audio_attributes_t &attributes,
|
|
const audio_config_base_t *config,
|
|
audio_input_flags_t flags,
|
|
const sp<AudioPolicyMix> &policyMix);
|
|
|
|
// event is one of STARTING_OUTPUT, STARTING_BEACON, STOPPING_OUTPUT, STOPPING_BEACON
|
|
// returns 0 if no mute/unmute event happened, the largest latency of the device where
|
|
// the mute/unmute happened
|
|
uint32_t handleEventForBeacon(int event);
|
|
uint32_t setBeaconMute(bool mute);
|
|
bool isValidAttributes(const audio_attributes_t *paa);
|
|
|
|
// Called by setDeviceConnectionState().
|
|
status_t setDeviceConnectionStateInt(audio_devices_t deviceType,
|
|
audio_policy_dev_state_t state,
|
|
const char *device_address,
|
|
const char *device_name,
|
|
audio_format_t encodedFormat);
|
|
status_t setDeviceConnectionStateInt(const sp<DeviceDescriptor> &device,
|
|
audio_policy_dev_state_t state);
|
|
|
|
void setEngineDeviceConnectionState(const sp<DeviceDescriptor> device,
|
|
audio_policy_dev_state_t state);
|
|
|
|
void updateMono(audio_io_handle_t output) {
|
|
AudioParameter param;
|
|
param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
|
|
mpClientInterface->setParameters(output, param.toString());
|
|
}
|
|
|
|
/**
|
|
* @brief createAudioPatchInternal internal function to manage audio patch creation
|
|
* @param[in] patch structure containing sink and source ports configuration
|
|
* @param[out] handle patch handle to be provided if patch installed correctly
|
|
* @param[in] uid of the client
|
|
* @param[in] delayMs if required
|
|
* @param[in] sourceDesc [optional] in case of external source, source client to be
|
|
* configured by the patch, i.e. assigning an Output (HW or SW)
|
|
* @return NO_ERROR if patch installed correctly, error code otherwise.
|
|
*/
|
|
status_t createAudioPatchInternal(const struct audio_patch *patch,
|
|
audio_patch_handle_t *handle,
|
|
uid_t uid, uint32_t delayMs = 0,
|
|
const sp<SourceClientDescriptor>& sourceDesc = nullptr);
|
|
/**
|
|
* @brief releaseAudioPatchInternal internal function to remove an audio patch
|
|
* @param[in] handle of the patch to be removed
|
|
* @param[in] delayMs if required
|
|
* @return NO_ERROR if patch removed correctly, error code otherwise.
|
|
*/
|
|
status_t releaseAudioPatchInternal(audio_patch_handle_t handle, uint32_t delayMs = 0);
|
|
|
|
status_t installPatch(const char *caller,
|
|
audio_patch_handle_t *patchHandle,
|
|
AudioIODescriptorInterface *ioDescriptor,
|
|
const struct audio_patch *patch,
|
|
int delayMs);
|
|
status_t installPatch(const char *caller,
|
|
ssize_t index,
|
|
audio_patch_handle_t *patchHandle,
|
|
const struct audio_patch *patch,
|
|
int delayMs,
|
|
uid_t uid,
|
|
sp<AudioPatch> *patchDescPtr);
|
|
|
|
bool areAllDevicesSupported(
|
|
const AudioDeviceTypeAddrVector& devices,
|
|
std::function<bool(audio_devices_t)> predicate,
|
|
const char* context);
|
|
|
|
bool isScoRequestedForComm() const;
|
|
|
|
bool areAllActiveTracksRerouted(const sp<SwAudioOutputDescriptor>& output);
|
|
|
|
sp<SwAudioOutputDescriptor> openOutputWithProfileAndDevice(const sp<IOProfile>& profile,
|
|
const DeviceVector& devices);
|
|
|
|
};
|
|
|
|
};
|