Refactor AudioPolicyManager
AudioPolicyManager implementation is now split into the
following files:
files managerdefault/Gains.*
class AudioGain
class VolumeCurvePoint
class StreamDescriptor
files managerdefault/Devices.*
class DeviceDescriptor
class DeviceVector
files managerdefault/Ports.*
class AudioPort
class AudioPortConfig
class AudioPatch
files managerdefault/IOProfile.*
class IOProfile
files managerdefault/HwModule.*
class HwModule
files managerdefault/AudioInputDescriptor.*
class AudioInputDescriptor
files managerdefault/AudioOutputDescriptor.*
class AudioOutputDescriptor
All files for libaudiopolicyservice are moved under service/
All files for libaudiopolicymanager are moved under manager/
Change-Id: I43758be1894e37d34db194b51a19ae24461e066e
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index 188fc89..351ed79 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -3,19 +3,19 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyService.cpp \
- AudioPolicyEffects.cpp
+ service/AudioPolicyService.cpp \
+ service/AudioPolicyEffects.cpp
ifeq ($(USE_LEGACY_AUDIO_POLICY), 1)
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImplLegacy.cpp \
- AudioPolicyClientImplLegacy.cpp
+ service/AudioPolicyInterfaceImplLegacy.cpp \
+ service/AudioPolicyClientImplLegacy.cpp
LOCAL_CFLAGS += -DUSE_LEGACY_AUDIO_POLICY
else
LOCAL_SRC_FILES += \
- AudioPolicyInterfaceImpl.cpp \
- AudioPolicyClientImpl.cpp
+ service/AudioPolicyInterfaceImpl.cpp \
+ service/AudioPolicyClientImpl.cpp
endif
LOCAL_C_INCLUDES := \
@@ -53,7 +53,15 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyManager.cpp
+ managerdefault/AudioPolicyManager.cpp \
+ managerdefault/ConfigParsingUtils.cpp \
+ managerdefault/Devices.cpp \
+ managerdefault/Gains.cpp \
+ managerdefault/HwModule.cpp \
+ managerdefault/IOProfile.cpp \
+ managerdefault/Ports.cpp \
+ managerdefault/AudioInputDescriptor.cpp \
+ managerdefault/AudioOutputDescriptor.cpp
LOCAL_SHARED_LIBRARIES := \
libcutils \
@@ -73,7 +81,7 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- AudioPolicyFactory.cpp
+ manager/AudioPolicyFactory.cpp
LOCAL_SHARED_LIBRARIES := \
libaudiopolicymanagerdefault
diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h
deleted file mode 100644
index 81d4f14..0000000
--- a/services/audiopolicy/AudioPolicyManager.h
+++ /dev/null
@@ -1,951 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <cutils/config_utils.h>
-#include <cutils/misc.h>
-#include <utils/Timers.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/SortedVector.h>
-#include <media/AudioPolicy.h>
-#include "AudioPolicyInterface.h"
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-// 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
-
-#define MAX_MIXER_SAMPLING_RATE 48000
-#define MAX_MIXER_CHANNEL_COUNT 8
-
-// ----------------------------------------------------------------------------
-// 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);
- virtual ~AudioPolicyManager();
-
- // AudioPolicyInterface
- virtual status_t setDeviceConnectionState(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address,
- const char *device_name);
- 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,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- virtual status_t getOutputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *output,
- audio_session_t session,
- audio_stream_type_t *stream,
- uint32_t samplingRate,
- 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,
- audio_session_t session);
- virtual status_t stopOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual void releaseOutput(audio_io_handle_t output,
- audio_stream_type_t stream,
- audio_session_t session);
- virtual status_t getInputForAttr(const audio_attributes_t *attr,
- audio_io_handle_t *input,
- audio_session_t session,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags,
- input_type_t *inputType);
-
- // indicates to the audio policy manager that the input starts being used.
- virtual status_t startInput(audio_io_handle_t input,
- audio_session_t session);
-
- // indicates to the audio policy manager that the input stops being used.
- virtual status_t stopInput(audio_io_handle_t input,
- audio_session_t session);
- virtual void releaseInput(audio_io_handle_t input,
- audio_session_t session);
- 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 strategy corresponding to the given audio attributes
- virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
-
- // 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
- // For the base implementation, "remotely" means playing during screen mirroring which
- // uses an output for playback with a non-empty, non "0" address.
- 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 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);
-
- virtual status_t acquireSoundTriggerSession(audio_session_t *session,
- audio_io_handle_t *ioHandle,
- audio_devices_t *device);
-
- virtual status_t releaseSoundTriggerSession(audio_session_t session);
-
- virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
- virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
-
-protected:
-
- enum routing_strategy {
- STRATEGY_MEDIA,
- STRATEGY_PHONE,
- STRATEGY_SONIFICATION,
- STRATEGY_SONIFICATION_RESPECTFUL,
- STRATEGY_DTMF,
- STRATEGY_ENFORCED_AUDIBLE,
- STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
- STRATEGY_ACCESSIBILITY,
- STRATEGY_REROUTING,
- 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_EXT_MEDIA,
- DEVICE_CATEGORY_CNT
- };
-
- class HwModule;
-
- class AudioGain: public RefBase
- {
- public:
- AudioGain(int index, bool useInChannelMask);
- virtual ~AudioGain() {}
-
- void dump(int fd, int spaces, int index) const;
-
- void getDefaultConfig(struct audio_gain_config *config);
- status_t checkConfig(const struct audio_gain_config *config);
- int mIndex;
- struct audio_gain mGain;
- bool mUseInChannelMask;
- };
-
- class AudioPort: public virtual RefBase
- {
- public:
- AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module);
- virtual ~AudioPort() {}
-
- audio_port_handle_t getHandle() { return mId; }
-
- void attach(const sp<HwModule>& module);
- bool isAttached() { return mId != 0; }
-
- virtual void toAudioPort(struct audio_port *port) const;
-
- void importAudioPort(const sp<AudioPort> port);
- void clearCapabilities();
-
- 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, int index);
- virtual void loadGains(cnode *root);
-
- // searches for an exact match
- status_t checkExactSamplingRate(uint32_t samplingRate) const;
- // searches for a compatible match, and returns the best match via updatedSamplingRate
- status_t checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const;
- // searches for an exact match
- status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
- // searches for a compatible match, currently implemented for input channel masks only
- status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
- status_t checkFormat(audio_format_t format) const;
- status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
- uint32_t pickSamplingRate() const;
- audio_channel_mask_t pickChannelMask() const;
- audio_format_t pickFormat() const;
-
- static const audio_format_t sPcmFormatCompareTable[];
- static int compareFormats(audio_format_t format1, audio_format_t format2);
-
- void dump(int fd, int spaces) const;
-
- String8 mName;
- audio_port_type_t mType;
- audio_port_role_t mRole;
- bool mUseInChannelMask;
- // 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
- sp<HwModule> mModule; // audio HW module exposing this I/O stream
- uint32_t mFlags; // attribute flags (e.g primary output,
- // direct output...).
-
- protected:
- //TODO - clarify the role of mId in this case, both an "attached" indicator
- // and a unique ID for identifying a port to the (upcoming) selection API,
- // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
- audio_port_handle_t mId;
- };
-
- class AudioPortConfig: public virtual RefBase
- {
- public:
- AudioPortConfig();
- virtual ~AudioPortConfig() {}
-
- status_t applyAudioPortConfig(const struct audio_port_config *config,
- struct audio_port_config *backupConfig = NULL);
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const = 0;
- virtual sp<AudioPort> getAudioPort() const = 0;
- uint32_t mSamplingRate;
- audio_format_t mFormat;
- audio_channel_mask_t mChannelMask;
- struct audio_gain_config mGain;
- };
-
-
- 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) {}
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_patch_handle_t mHandle;
- struct audio_patch mPatch;
- uid_t mUid;
- audio_patch_handle_t mAfPatchHandle;
- };
-
- class DeviceDescriptor: public AudioPort, public AudioPortConfig
- {
- public:
- DeviceDescriptor(const String8& name, audio_devices_t type);
-
- virtual ~DeviceDescriptor() {}
-
- bool equals(const sp<DeviceDescriptor>& other) const;
-
- // AudioPortConfig
- virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
-
- // AudioPort
- virtual void loadGains(cnode *root);
- virtual void toAudioPort(struct audio_port *port) const;
-
- status_t dump(int fd, int spaces, int index) const;
-
- audio_devices_t mDeviceType;
- String8 mAddress;
-
- static String8 emptyNameStr;
- };
-
- 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;
- DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
- 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, const sp<HwModule>& module);
- virtual ~IOProfile();
-
- // This method is used for both output and input.
- // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
- // For input, flags is interpreted as audio_input_flags_t.
- // TODO: merge audio_output_flags_t and audio_input_flags_t.
- bool isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const;
-
- void dump(int fd);
- void log();
-
- DeviceVector mSupportedDevices; // supported devices
- // (devices this output can be routed to)
- };
-
- class HwModule : public RefBase
- {
- public:
- HwModule(const char *name);
- ~HwModule();
-
- status_t loadOutput(cnode *root);
- status_t loadInput(cnode *root);
- status_t loadDevice(cnode *root);
-
- status_t addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeOutputProfile(String8 name);
- status_t addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address);
- status_t removeInputProfile(String8 name);
-
- void dump(int fd);
-
- const char *const mName; // base name of the audio HW module (primary, a2dp ...)
- uint32_t mHalVersion; // audio HAL API version
- 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 non-media audio on ext media outputs (HDMI, Line, etc)
- static const VolumeCurvePoint sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT];
- // volume curve for media strategy on speakers
- static const VolumeCurvePoint sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[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];
- static const VolumeCurvePoint sLinearVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sSilentVolumeCurve[AudioPolicyManager::VOLCNT];
- static const VolumeCurvePoint sFullScaleVolumeCurve[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 AudioPortConfig
- {
- 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 sp<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;
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- void toAudioPort(struct audio_port *port) const;
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // output handle
- uint32_t mLatency; //
- audio_output_flags_t mFlags; //
- audio_devices_t mDevice; // current device this output is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- 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];
- sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
- sp<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 AudioPortConfig
- {
- public:
- AudioInputDescriptor(const sp<IOProfile>& profile);
-
- status_t dump(int fd);
-
- audio_port_handle_t mId;
- audio_io_handle_t mIoHandle; // input handle
- audio_devices_t mDevice; // current device this input is routed to
- AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
- audio_patch_handle_t mPatchHandle;
- uint32_t mRefCount; // number of AudioRecord clients using
- // this input
- uint32_t mOpenRefCount;
- audio_source_t mInputSource; // input source selected by application
- //(mediarecorder.h)
- const sp<IOProfile> mProfile; // I/O profile this output derives from
- SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
- bool mIsSoundTrigger; // used by a soundtrigger capture
-
- virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig = NULL) const;
- virtual sp<AudioPort> getAudioPort() const { return mProfile; }
- 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 RefBase
- {
- 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, sp<AudioOutputDescriptor> outputDesc);
- void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
-
- // return the strategy corresponding to a given stream type
- 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...
- // if fromCache is true, the device is returned from mDeviceForStrategy[],
- // otherwise it is determine by current state
- // (device connected,phone state, force use, a2dp output...)
- // This allows to:
- // 1 speed up process when the state is stable (when starting or stopping an output)
- // 2 access to either current device selection (fromCache == true) or
- // "future" device selection (fromCache == false) when called from a context
- // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
- // before updateDevicesAndOutputs() is called.
- virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
- bool fromCache);
-
- // change the route of the specified output. Returns the number of ms we have slept to
- // allow new routing to take effect in certain cases.
- virtual uint32_t setOutputDevice(audio_io_handle_t output,
- audio_devices_t device,
- bool force = false,
- int delayMs = 0,
- audio_patch_handle_t *patchHandle = NULL,
- const char* address = 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(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);
-
- uint32_t activeInputsCount() const;
-
- // 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(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
- virtual 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(const sp<DeviceDescriptor> devDesc,
- 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);
-
- // close an input.
- void closeInput(audio_io_handle_t input);
-
- // 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(const sp<EffectDescriptor>& effectDesc, bool enabled);
-
- // returns the category the device belongs to with regard to volume curve management
- static device_category getDeviceCategory(audio_devices_t device);
-
- // extract one device relevant for volume control from multiple device selection
- static audio_devices_t getDeviceForVolume(audio_devices_t device);
-
- SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
- bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
- SortedVector<audio_io_handle_t>& outputs2);
-
- // mute/unmute strategies using an incompatible device combination
- // if muting, wait for the audio in pcm buffer to be drained before proceeding
- // if unmuting, unmute only after the specified delay
- // Returns the number of ms waited
- virtual uint32_t checkDeviceMuteStrategies(sp<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,
- audio_format_t format);
- // samplingRate parameter is an in/out and so may be modified
- sp<IOProfile> getInputProfile(audio_devices_t device,
- String8 address,
- uint32_t& samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_input_flags_t flags);
- 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();
-
- virtual status_t addAudioPatch(audio_patch_handle_t handle,
- const sp<AudioPatch>& patch);
- virtual status_t removeAudioPatch(audio_patch_handle_t handle);
-
- sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
- sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
- sp<HwModule> getModuleForDevice(audio_devices_t device) const;
- sp<HwModule> getModuleFromName(const char *name) const;
- audio_devices_t availablePrimaryOutputDevices();
- audio_devices_t availablePrimaryInputDevices();
-
- void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
-
- //
- // 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 uint32_t parseOutputFlagNames(char *name);
- static uint32_t parseInputFlagNames(char *name);
- static audio_devices_t parseDeviceNames(char *name);
- void loadHwModule(cnode *root);
- void loadHwModules(cnode *root);
- void loadGlobalConfig(cnode *root, const sp<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, sp<AudioOutputDescriptor> > mOutputs;
- // copy of mOutputs before setDeviceConnectionState() opens new outputs
- // reset to mOutputs when updateDevicesAndOutputs() is called.
- DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
- DefaultKeyedVector<audio_io_handle_t, sp<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, sp<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 < sp<HwModule> > mHwModules;
- static volatile int32_t mNextUniqueId;
- volatile int32_t mAudioPortGeneration;
-
- DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
-
- DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
-
- sp<AudioPatch> mCallTxPatch;
- sp<AudioPatch> mCallRxPatch;
-
- // 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
-
- // custom mix entry in mPolicyMixes
- class AudioPolicyMix : public RefBase {
- public:
- AudioPolicyMix() {}
-
- AudioMix mMix; // Audio policy mix descriptor
- sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
- };
- DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
-
-
-#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
- static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi);
- static bool isVirtualInputDevice(audio_devices_t device);
- static uint32_t nextUniqueId();
- uint32_t nextAudioPortGeneration();
-private:
- // updates device caching and output for streams that can influence the
- // routing of notifications
- void handleNotificationRoutingForStream(audio_stream_type_t stream);
- static bool deviceDistinguishesOnAddress(audio_devices_t device);
- // find the outputs on a given output descriptor that have the given address.
- // to be called on an AudioOutputDescriptor whose supported devices (as defined
- // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
- // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
- // where addresses are used to distinguish between one connected device and another.
- void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
- const audio_devices_t device /*in*/,
- const String8 address /*in*/,
- SortedVector<audio_io_handle_t>& outputs /*out*/);
- uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
- // internal method to return the output handle for the given device and format
- audio_io_handle_t getOutputForDevice(
- audio_devices_t device,
- audio_session_t session,
- audio_stream_type_t stream,
- uint32_t samplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- audio_output_flags_t flags,
- const audio_offload_info_t *offloadInfo);
- // internal function to derive a stream type value from audio attributes
- audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
- // return true if any output is playing anything besides the stream to ignore
- bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
- // 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);
-
- // select input device corresponding to requested audio source and return associated policy
- // mix if any. Calls getDeviceForInputSource().
- audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
- AudioMix **policyMix = NULL);
-
- // Called by setDeviceConnectionState().
- status_t setDeviceConnectionStateInt(audio_devices_t device,
- audio_policy_dev_state_t state,
- const char *device_address,
- const char *device_name);
- sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
- const char *device_address,
- const char *device_name);
-};
-
-};
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
similarity index 94%
rename from services/audiopolicy/AudioPolicyFactory.cpp
rename to services/audiopolicy/manager/AudioPolicyFactory.cpp
index 2ae7bc1..9910a1f 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {
diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
similarity index 60%
copy from services/audiopolicy/AudioPolicyFactory.cpp
copy to services/audiopolicy/managerdefault/ApmImplDefinitions.h
index 2ae7bc1..620979b 100644
--- a/services/audiopolicy/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/managerdefault/ApmImplDefinitions.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2014 The Android Open Source Project
+ * Copyright (C) 2015 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,19 +14,19 @@
* limitations under the License.
*/
-#include "AudioPolicyManager.h"
-
namespace android {
-extern "C" AudioPolicyInterface* createAudioPolicyManager(
- AudioPolicyClientInterface *clientInterface)
-{
- return new AudioPolicyManager(clientInterface);
-}
+enum routing_strategy {
+ STRATEGY_MEDIA,
+ STRATEGY_PHONE,
+ STRATEGY_SONIFICATION,
+ STRATEGY_SONIFICATION_RESPECTFUL,
+ STRATEGY_DTMF,
+ STRATEGY_ENFORCED_AUDIBLE,
+ STRATEGY_TRANSMITTED_THROUGH_SPEAKER,
+ STRATEGY_ACCESSIBILITY,
+ STRATEGY_REROUTING,
+ NUM_STRATEGIES
+};
-extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
-{
- delete interface;
-}
-
-}; // namespace android
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
new file mode 100644
index 0000000..f4054c8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.cpp
@@ -0,0 +1,100 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioInputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0),
+ mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
+ mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
+{
+ if (profile != NULL) {
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+void AudioInputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(mProfile != 0,
+ "toAudioPortConfig() called on input with null profile %d", mIoHandle);
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SINK;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.source = mInputSource;
+}
+
+void AudioInputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
+
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioInputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %d\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioInputDescriptor.h b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
new file mode 100644
index 0000000..02579e6
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioInputDescriptor.h
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+// 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 AudioPortConfig
+{
+public:
+ AudioInputDescriptor(const sp<IOProfile>& profile);
+
+ status_t dump(int fd);
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // input handle
+ audio_devices_t mDevice; // current device this input is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ audio_patch_handle_t mPatchHandle;
+ uint32_t mRefCount; // number of AudioRecord clients using
+ // this input
+ uint32_t mOpenRefCount;
+ audio_source_t mInputSource; // input source selected by application
+ //(mediarecorder.h)
+ const sp<IOProfile> mProfile; // I/O profile this output derives from
+ SortedVector<audio_session_t> mSessions; // audio sessions attached to this input
+ bool mIsSoundTrigger; // used by a soundtrigger capture
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
new file mode 100644
index 0000000..4b85972
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.cpp
@@ -0,0 +1,221 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::AudioOutputDescriptor"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+AudioOutputDescriptor::AudioOutputDescriptor(
+ const sp<IOProfile>& profile)
+ : mId(0), mIoHandle(0), mLatency(0),
+ mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
+ mPatchHandle(0),
+ mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
+{
+ // clear usage count for all stream types
+ for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
+ mRefCount[i] = 0;
+ mCurVolume[i] = -1.0;
+ mMuteCount[i] = 0;
+ mStopTime[i] = 0;
+ }
+ for (int i = 0; i < NUM_STRATEGIES; i++) {
+ mStrategyMutedByDevice[i] = false;
+ }
+ if (profile != NULL) {
+ mFlags = (audio_output_flags_t)profile->mFlags;
+ mSamplingRate = profile->pickSamplingRate();
+ mFormat = profile->pickFormat();
+ mChannelMask = profile->pickChannelMask();
+ if (profile->mGains.size() > 0) {
+ profile->mGains[0]->getDefaultConfig(&mGain);
+ }
+ }
+}
+
+audio_devices_t AudioOutputDescriptor::device() const
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
+ } else {
+ return mDevice;
+ }
+}
+
+uint32_t AudioOutputDescriptor::latency()
+{
+ if (isDuplicated()) {
+ return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
+ } else {
+ return mLatency;
+ }
+}
+
+bool AudioOutputDescriptor::sharesHwModuleWith(
+ const sp<AudioOutputDescriptor> outputDesc)
+{
+ if (isDuplicated()) {
+ return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
+ } else if (outputDesc->isDuplicated()){
+ return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
+ } else {
+ return (mProfile->mModule == outputDesc->mProfile->mModule);
+ }
+}
+
+void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
+ int delta)
+{
+ // forward usage count change to attached outputs
+ if (isDuplicated()) {
+ mOutput1->changeRefCount(stream, delta);
+ mOutput2->changeRefCount(stream, delta);
+ }
+ if ((delta + (int)mRefCount[stream]) < 0) {
+ ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
+ delta, stream, mRefCount[stream]);
+ mRefCount[stream] = 0;
+ return;
+ }
+ mRefCount[stream] += delta;
+ ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
+}
+
+audio_devices_t AudioOutputDescriptor::supportedDevices()
+{
+ if (isDuplicated()) {
+ return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
+ } else {
+ return mProfile->mSupportedDevices.types() ;
+ }
+}
+
+bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const
+{
+ return isStrategyActive(NUM_STRATEGIES, inPastMs);
+}
+
+bool AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if ((sysTime == 0) && (inPastMs != 0)) {
+ sysTime = systemTime();
+ }
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ if (i == AUDIO_STREAM_PATCH) {
+ continue;
+ }
+ if (((AudioPolicyManager::getStrategy((audio_stream_type_t)i) == strategy) ||
+ (NUM_STRATEGIES == strategy)) &&
+ isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
+ return true;
+ }
+ }
+ return false;
+}
+
+bool AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
+ uint32_t inPastMs,
+ nsecs_t sysTime) const
+{
+ if (mRefCount[stream] != 0) {
+ return true;
+ }
+ if (inPastMs == 0) {
+ return false;
+ }
+ if (sysTime == 0) {
+ sysTime = systemTime();
+ }
+ if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
+ return true;
+ }
+ return false;
+}
+
+void AudioOutputDescriptor::toAudioPortConfig(
+ struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
+
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
+ AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_MIX;
+ dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
+ dstConfig->ext.mix.handle = mIoHandle;
+ dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+}
+
+void AudioOutputDescriptor::toAudioPort(
+ struct audio_port *port) const
+{
+ ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
+ mProfile->toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.mix.hw_module = mProfile->mModule->mHandle;
+ port->ext.mix.handle = mIoHandle;
+ port->ext.mix.latency_class =
+ mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
+}
+
+status_t AudioOutputDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " ID: %d\n", mId);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Devices %08x\n", device());
+ result.append(buffer);
+ snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
+ result.append(buffer);
+ for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
+ snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
+ i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+
+ return NO_ERROR;
+}
+
+
+
+}; //namespace android
diff --git a/services/audiopolicy/managerdefault/AudioOutputDescriptor.h b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
new file mode 100644
index 0000000..32f46e4
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioOutputDescriptor.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include "ApmImplDefinitions.h"
+
+namespace android {
+
+// 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 AudioPortConfig
+{
+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 sp<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;
+
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+ virtual sp<AudioPort> getAudioPort() const { return mProfile; }
+ void toAudioPort(struct audio_port *port) const;
+
+ audio_port_handle_t mId;
+ audio_io_handle_t mIoHandle; // output handle
+ uint32_t mLatency; //
+ audio_output_flags_t mFlags; //
+ audio_devices_t mDevice; // current device this output is routed to
+ AudioMix *mPolicyMix; // non NULL when used by a dynamic policy
+ 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];
+ sp<AudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output
+ sp<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)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
similarity index 73%
rename from services/audiopolicy/AudioPolicyManager.cpp
rename to services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 50ea6ff..b48dc80 100644
--- a/services/audiopolicy/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -14,7 +14,7 @@
* limitations under the License.
*/
-#define LOG_TAG "AudioPolicyManager"
+#define LOG_TAG "APM::AudioPolicyManager"
//#define LOG_NDEBUG 0
//#define VERY_VERBOSE_LOGGING
@@ -51,163 +51,6 @@
namespace android {
// ----------------------------------------------------------------------------
-// Definitions for audio_policy.conf file parsing
-// ----------------------------------------------------------------------------
-
-struct StringToEnum {
- const char *name;
- uint32_t value;
-};
-
-#define STRING_TO_ENUM(string) { #string, string }
-#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
-
-const StringToEnum sDeviceNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
- STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
- STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
-};
-
-const StringToEnum sOutputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
- STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
-};
-
-const StringToEnum sInputFlagNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
- STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
-};
-
-const StringToEnum sFormatNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
- STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
- STRING_TO_ENUM(AUDIO_FORMAT_MP3),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
- STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
- STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
- STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
- STRING_TO_ENUM(AUDIO_FORMAT_AC3),
- STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
-};
-
-const StringToEnum sOutChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
- STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
-};
-
-const StringToEnum sInChannelsNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
- STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
-};
-
-const StringToEnum sGainModeNameToEnumTable[] = {
- STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
- STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
-};
-
-
-uint32_t AudioPolicyManager::stringToEnum(const struct StringToEnum *table,
- size_t size,
- const char *name)
-{
- for (size_t i = 0; i < size; i++) {
- if (strcmp(table[i].name, name) == 0) {
- ALOGV("stringToEnum() found %s", table[i].name);
- return table[i].value;
- }
- }
- return 0;
-}
-
-const char *AudioPolicyManager::enumToString(const struct StringToEnum *table,
- size_t size,
- uint32_t value)
-{
- for (size_t i = 0; i < size; i++) {
- if (table[i].value == value) {
- return table[i].name;
- }
- }
- return "";
-}
-
-bool AudioPolicyManager::stringToBool(const char *value)
-{
- return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
-}
-
-
-// ----------------------------------------------------------------------------
// AudioPolicyInterface implementation
// ----------------------------------------------------------------------------
@@ -276,8 +119,7 @@
ALOGV("setDeviceConnectionState() checkOutputsForDevice() returned %zu outputs",
outputs.size());
-
- // Set connect to HALs
+ // Send connect to HALs
AudioParameter param = AudioParameter(devDesc->mAddress);
param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device);
mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
@@ -452,10 +294,9 @@
}
}
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(
- const audio_devices_t device,
- const char *device_address,
- const char *device_name)
+sp<DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name)
{
String8 address = (device_address == NULL) ? String8("") : String8(device_address);
// handle legacy remote submix case where the address was not always specified
@@ -642,18 +483,18 @@
// force routing command to audio hardware when starting a call
// even if no device change is needed
force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j];
}
} else if (isStateInCall(oldState) && !isStateInCall(state)) {
ALOGV(" Exiting call in setPhoneState()");
// force routing command to audio hardware when exiting a call
// even if no device change is needed
force = true;
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] =
- sVolumeProfiles[AUDIO_STREAM_DTMF][j];
+ ApmGains::sVolumeProfiles[AUDIO_STREAM_DTMF][j];
}
} else if (isStateInCall(state) && (state != oldState)) {
ALOGV(" Switching between telephony and VoIP in setPhoneState()");
@@ -844,7 +685,7 @@
// Find a direct output profile compatible with the parameters passed, even if the input flags do
// not explicitly request a direct output
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput(
+sp<IOProfile> AudioPolicyManager::getProfileForDirectOutput(
audio_devices_t device,
uint32_t samplingRate,
audio_format_t format,
@@ -1132,6 +973,10 @@
if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
goto non_direct_output;
}
+ // fall back to mixer output if possible when the direct output could not be open
+ if (audio_is_linear_pcm(format) && samplingRate <= MAX_MIXER_SAMPLING_RATE) {
+ goto non_direct_output;
+ }
return AUDIO_IO_HANDLE_NONE;
}
outputDesc->mSamplingRate = config.sample_rate;
@@ -1853,7 +1698,7 @@
status_t status = NO_ERROR;
for (size_t i = 0; i < mOutputs.size(); i++) {
audio_devices_t curDevice =
- getDeviceForVolume(mOutputs.valueAt(i)->device());
+ ApmGains::getDeviceForVolume(mOutputs.valueAt(i)->device());
if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) {
status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice);
if (volStatus != NO_ERROR) {
@@ -1883,7 +1728,7 @@
if (device == AUDIO_DEVICE_OUT_DEFAULT) {
device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/);
}
- device = getDeviceForVolume(device);
+ device = ApmGains::getDeviceForVolume(device);
*index = mStreams[stream].getVolumeIndex(device);
ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index);
@@ -2467,7 +2312,7 @@
return NO_ERROR;
}
-sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
+sp<AudioOutputDescriptor> AudioPolicyManager::getOutputFromId(
audio_port_handle_t id) const
{
sp<AudioOutputDescriptor> outputDesc = NULL;
@@ -2480,7 +2325,7 @@
return outputDesc;
}
-sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId(
+sp<AudioInputDescriptor> AudioPolicyManager::getInputFromId(
audio_port_handle_t id) const
{
sp<AudioInputDescriptor> inputDesc = NULL;
@@ -2493,7 +2338,7 @@
return inputDesc;
}
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice(
+sp <HwModule> AudioPolicyManager::getModuleForDevice(
audio_devices_t device) const
{
sp <HwModule> module;
@@ -2521,7 +2366,7 @@
return module;
}
-sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
+sp <HwModule> AudioPolicyManager::getModuleFromName(const char *name) const
{
sp <HwModule> module;
@@ -4332,7 +4177,7 @@
return AUDIO_DEVICE_NONE;
}
audio_devices_t devices;
- AudioPolicyManager::routing_strategy strategy = getStrategy(stream);
+ routing_strategy strategy = getStrategy(stream);
devices = getDeviceForStrategy(strategy, true /*fromCache*/);
SortedVector<audio_io_handle_t> outputs = getOutputsForDevice(devices, mOutputs);
for (size_t i = 0; i < outputs.size(); i++) {
@@ -4353,7 +4198,7 @@
return devices;
}
-AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy(
+routing_strategy AudioPolicyManager::getStrategy(
audio_stream_type_t stream) {
ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH");
@@ -5132,7 +4977,7 @@
return status;
}
-sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
+sp<IOProfile> AudioPolicyManager::getInputProfile(audio_devices_t device,
String8 address,
uint32_t& samplingRate,
audio_format_t format,
@@ -5342,305 +5187,29 @@
}
-audio_devices_t AudioPolicyManager::getDeviceForVolume(audio_devices_t device)
-{
- if (device == AUDIO_DEVICE_NONE) {
- // this happens when forcing a route update and no track is active on an output.
- // In this case the returned category is not important.
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (popcount(device) > 1) {
- // Multiple device selection is either:
- // - speaker + one other device: give priority to speaker in this case.
- // - one A2DP device + another device: happens with duplicated output. In this case
- // retain the device on the A2DP output as the other must not correspond to an active
- // selection if not the speaker.
- // - HDMI-CEC system audio mode only output: give priority to available item in order.
- if (device & AUDIO_DEVICE_OUT_SPEAKER) {
- device = AUDIO_DEVICE_OUT_SPEAKER;
- } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
- device = AUDIO_DEVICE_OUT_HDMI_ARC;
- } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
- device = AUDIO_DEVICE_OUT_AUX_LINE;
- } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
- device = AUDIO_DEVICE_OUT_SPDIF;
- } else {
- device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
- }
- }
-
- /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
- if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
- device = AUDIO_DEVICE_OUT_SPEAKER;
-
- ALOGW_IF(popcount(device) != 1,
- "getDeviceForVolume() invalid device combination: %08x",
- device);
-
- return device;
-}
-
-AudioPolicyManager::device_category AudioPolicyManager::getDeviceCategory(audio_devices_t device)
-{
- switch(getDeviceForVolume(device)) {
- case AUDIO_DEVICE_OUT_EARPIECE:
- return DEVICE_CATEGORY_EARPIECE;
- case AUDIO_DEVICE_OUT_WIRED_HEADSET:
- case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
- return DEVICE_CATEGORY_HEADSET;
- case AUDIO_DEVICE_OUT_LINE:
- case AUDIO_DEVICE_OUT_AUX_DIGITAL:
- /*USB? Remote submix?*/
- return DEVICE_CATEGORY_EXT_MEDIA;
- case AUDIO_DEVICE_OUT_SPEAKER:
- case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
- case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
- case AUDIO_DEVICE_OUT_USB_ACCESSORY:
- case AUDIO_DEVICE_OUT_USB_DEVICE:
- case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
- default:
- return DEVICE_CATEGORY_SPEAKER;
- }
-}
-
-/* static */
-float AudioPolicyManager::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
- int indexInUi)
-{
- device_category deviceCategory = getDeviceCategory(device);
- const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
-
- // the volume index in the UI is relative to the min and max volume indices for this stream type
- int nbSteps = 1 + curve[VOLMAX].mIndex -
- curve[VOLMIN].mIndex;
- int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
- (streamDesc.mIndexMax - streamDesc.mIndexMin);
-
- // find what part of the curve this index volume belongs to, or if it's out of bounds
- int segment = 0;
- if (volIdx < curve[VOLMIN].mIndex) { // out of bounds
- return 0.0f;
- } else if (volIdx < curve[VOLKNEE1].mIndex) {
- segment = 0;
- } else if (volIdx < curve[VOLKNEE2].mIndex) {
- segment = 1;
- } else if (volIdx <= curve[VOLMAX].mIndex) {
- segment = 2;
- } else { // out of bounds
- return 1.0f;
- }
-
- // linear interpolation in the attenuation table in dB
- float decibels = curve[segment].mDBAttenuation +
- ((float)(volIdx - curve[segment].mIndex)) *
- ( (curve[segment+1].mDBAttenuation -
- curve[segment].mDBAttenuation) /
- ((float)(curve[segment+1].mIndex -
- curve[segment].mIndex)) );
-
- float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
-
- ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
- curve[segment].mIndex, volIdx,
- curve[segment+1].mIndex,
- curve[segment].mDBAttenuation,
- decibels,
- curve[segment+1].mDBAttenuation,
- amplification);
-
- return amplification;
-}
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sExtMediaSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerMediaVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerSonificationVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
-};
-
-// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
-// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
-// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
-// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultSystemVolumeCurveDrc[AudioPolicyManager::VOLCNT] = {
- {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sHeadsetSystemVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sDefaultVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSpeakerVoiceVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sLinearVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sSilentVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- AudioPolicyManager::sFullScaleVolumeCurve[AudioPolicyManager::VOLCNT] = {
- {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
-};
-
-const AudioPolicyManager::VolumeCurvePoint
- *AudioPolicyManager::sVolumeProfiles[AUDIO_STREAM_CNT]
- [AudioPolicyManager::DEVICE_CATEGORY_CNT] = {
- { // AUDIO_STREAM_VOICE_CALL
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_SYSTEM
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_RING
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_MUSIC
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ALARM
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_NOTIFICATION
- sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_BLUETOOTH_SCO
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ENFORCED_AUDIBLE
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_DTMF
- sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_TTS
- // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_ACCESSIBILITY
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_REROUTING
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
- { // AUDIO_STREAM_PATCH
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
- sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
- sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
- },
-};
-
void AudioPolicyManager::initializeVolumeCurves()
{
for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- for (int j = 0; j < DEVICE_CATEGORY_CNT; j++) {
+ for (int j = 0; j < ApmGains::DEVICE_CATEGORY_CNT; j++) {
mStreams[i].mVolumeCurve[j] =
- sVolumeProfiles[i][j];
+ ApmGains::sVolumeProfiles[i][j];
}
}
// Check availability of DRC on speaker path: if available, override some of the speaker curves
if (mSpeakerDrcEnabled) {
- mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sDefaultSystemVolumeCurveDrc;
- mStreams[AUDIO_STREAM_RING].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerSonificationVolumeCurveDrc;
- mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
- mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[DEVICE_CATEGORY_SPEAKER] =
- sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_SYSTEM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sDefaultSystemVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_RING].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ALARM].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_NOTIFICATION].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerSonificationVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_MUSIC].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
+ mStreams[AUDIO_STREAM_ACCESSIBILITY].mVolumeCurve[ApmGains::DEVICE_CATEGORY_SPEAKER] =
+ ApmGains::sSpeakerMediaVolumeCurveDrc;
}
}
@@ -5657,7 +5226,7 @@
device = outputDesc->device();
}
- volume = volIndexToAmpl(device, streamDesc, index);
+ volume = ApmGains::volIndexToAmpl(device, streamDesc, index);
// if a headset is connected, apply the following rules to ring tones and notifications
// to avoid sound level bursts in user's ears:
@@ -5913,319 +5482,6 @@
}
-// --- AudioOutputDescriptor class implementation
-
-AudioPolicyManager::AudioOutputDescriptor::AudioOutputDescriptor(
- const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0), mLatency(0),
- mFlags((audio_output_flags_t)0), mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
- mPatchHandle(0),
- mOutput1(0), mOutput2(0), mProfile(profile), mDirectOpenCount(0)
-{
- // clear usage count for all stream types
- for (int i = 0; i < AUDIO_STREAM_CNT; i++) {
- mRefCount[i] = 0;
- mCurVolume[i] = -1.0;
- mMuteCount[i] = 0;
- mStopTime[i] = 0;
- }
- for (int i = 0; i < NUM_STRATEGIES; i++) {
- mStrategyMutedByDevice[i] = false;
- }
- if (profile != NULL) {
- mFlags = (audio_output_flags_t)profile->mFlags;
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::device() const
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice);
- } else {
- return mDevice;
- }
-}
-
-uint32_t AudioPolicyManager::AudioOutputDescriptor::latency()
-{
- if (isDuplicated()) {
- return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency;
- } else {
- return mLatency;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::sharesHwModuleWith(
- const sp<AudioOutputDescriptor> outputDesc)
-{
- if (isDuplicated()) {
- return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc);
- } else if (outputDesc->isDuplicated()){
- return sharesHwModuleWith(outputDesc->mOutput1) || sharesHwModuleWith(outputDesc->mOutput2);
- } else {
- return (mProfile->mModule == outputDesc->mProfile->mModule);
- }
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream,
- int delta)
-{
- // forward usage count change to attached outputs
- if (isDuplicated()) {
- mOutput1->changeRefCount(stream, delta);
- mOutput2->changeRefCount(stream, delta);
- }
- if ((delta + (int)mRefCount[stream]) < 0) {
- ALOGW("changeRefCount() invalid delta %d for stream %d, refCount %d",
- delta, stream, mRefCount[stream]);
- mRefCount[stream] = 0;
- return;
- }
- mRefCount[stream] += delta;
- ALOGV("changeRefCount() stream %d, count %d", stream, mRefCount[stream]);
-}
-
-audio_devices_t AudioPolicyManager::AudioOutputDescriptor::supportedDevices()
-{
- if (isDuplicated()) {
- return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices());
- } else {
- return mProfile->mSupportedDevices.types() ;
- }
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const
-{
- return isStrategyActive(NUM_STRATEGIES, inPastMs);
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStrategyActive(routing_strategy strategy,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if ((sysTime == 0) && (inPastMs != 0)) {
- sysTime = systemTime();
- }
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- if (i == AUDIO_STREAM_PATCH) {
- continue;
- }
- if (((getStrategy((audio_stream_type_t)i) == strategy) ||
- (NUM_STRATEGIES == strategy)) &&
- isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) {
- return true;
- }
- }
- return false;
-}
-
-bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream,
- uint32_t inPastMs,
- nsecs_t sysTime) const
-{
- if (mRefCount[stream] != 0) {
- return true;
- }
- if (inPastMs == 0) {
- return false;
- }
- if (sysTime == 0) {
- sysTime = systemTime();
- }
- if (ns2ms(sysTime - mStopTime[stream]) < inPastMs) {
- return true;
- }
- return false;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle);
-
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
-}
-
-void AudioPolicyManager::AudioOutputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle);
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class =
- mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioOutputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %08x\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Latency: %d\n", mLatency);
- result.append(buffer);
- snprintf(buffer, SIZE, " Flags %08x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", device());
- result.append(buffer);
- snprintf(buffer, SIZE, " Stream volume refCount muteCount\n");
- result.append(buffer);
- for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) {
- snprintf(buffer, SIZE, " %02d %.03f %02d %02d\n",
- i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- AudioInputDescriptor class implementation
-
-AudioPolicyManager::AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile)
- : mId(0), mIoHandle(0),
- mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0),
- mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false)
-{
- if (profile != NULL) {
- mSamplingRate = profile->pickSamplingRate();
- mFormat = profile->pickFormat();
- mChannelMask = profile->pickChannelMask();
- if (profile->mGains.size() > 0) {
- profile->mGains[0]->getDefaultConfig(&mGain);
- }
- }
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- ALOG_ASSERT(mProfile != 0,
- "toAudioPortConfig() called on input with null profile %d", mIoHandle);
- dstConfig->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK|
- AUDIO_PORT_CONFIG_FORMAT|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
-
- dstConfig->id = mId;
- dstConfig->role = AUDIO_PORT_ROLE_SINK;
- dstConfig->type = AUDIO_PORT_TYPE_MIX;
- dstConfig->ext.mix.hw_module = mProfile->mModule->mHandle;
- dstConfig->ext.mix.handle = mIoHandle;
- dstConfig->ext.mix.usecase.source = mInputSource;
-}
-
-void AudioPolicyManager::AudioInputDescriptor::toAudioPort(
- struct audio_port *port) const
-{
- ALOG_ASSERT(mProfile != 0, "toAudioPort() called on input with null profile %d", mIoHandle);
-
- mProfile->toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.mix.hw_module = mProfile->mModule->mHandle;
- port->ext.mix.handle = mIoHandle;
- port->ext.mix.latency_class = AUDIO_LATENCY_NORMAL;
-}
-
-status_t AudioPolicyManager::AudioInputDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " ID: %d\n", mId);
- result.append(buffer);
- snprintf(buffer, SIZE, " Sampling rate: %d\n", mSamplingRate);
- result.append(buffer);
- snprintf(buffer, SIZE, " Format: %d\n", mFormat);
- result.append(buffer);
- snprintf(buffer, SIZE, " Channels: %08x\n", mChannelMask);
- result.append(buffer);
- snprintf(buffer, SIZE, " Devices %08x\n", mDevice);
- result.append(buffer);
- snprintf(buffer, SIZE, " Ref Count %d\n", mRefCount);
- result.append(buffer);
- snprintf(buffer, SIZE, " Open Ref Count %d\n", mOpenRefCount);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-
- return NO_ERROR;
-}
-
-// --- StreamDescriptor class implementation
-
-AudioPolicyManager::StreamDescriptor::StreamDescriptor()
- : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
-{
- mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
-}
-
-int AudioPolicyManager::StreamDescriptor::getVolumeIndex(audio_devices_t device)
-{
- device = AudioPolicyManager::getDeviceForVolume(device);
- // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
- if (mIndexCur.indexOfKey(device) < 0) {
- device = AUDIO_DEVICE_OUT_DEFAULT;
- }
- return mIndexCur.valueFor(device);
-}
-
-void AudioPolicyManager::StreamDescriptor::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%s %02d %02d ",
- mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
- result.append(buffer);
- for (size_t i = 0; i < mIndexCur.size(); i++) {
- snprintf(buffer, SIZE, "%04x : %02d, ",
- mIndexCur.keyAt(i),
- mIndexCur.valueAt(i));
- result.append(buffer);
- }
- result.append("\n");
-
- write(fd, result.string(), result.size());
-}
-
// --- EffectDescriptor class implementation
status_t AudioPolicyManager::EffectDescriptor::dump(int fd)
@@ -6249,1611 +5505,9 @@
return NO_ERROR;
}
-// --- HwModule class implementation
-
-AudioPolicyManager::HwModule::HwModule(const char *name)
- : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
- mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
-{
-}
-
-AudioPolicyManager::HwModule::~HwModule()
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- mOutputProfiles[i]->mSupportedDevices.clear();
- }
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- mInputProfiles[i]->mSupportedDevices.clear();
- }
- free((void *)mName);
-}
-
-status_t AudioPolicyManager::HwModule::loadInput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadInChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseInputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadInput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadInput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadInput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadInput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadInput() adding input Supported Devices %04x",
- profile->mSupportedDevices.types());
-
- mInputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadOutput(cnode *root)
-{
- cnode *node = root->first_child;
-
- sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
-
- while (node) {
- if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
- profile->loadSamplingRates((char *)node->value);
- } else if (strcmp(node->name, FORMATS_TAG) == 0) {
- profile->loadFormats((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- profile->loadOutChannels((char *)node->value);
- } else if (strcmp(node->name, DEVICES_TAG) == 0) {
- profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
- mDeclaredDevices);
- } else if (strcmp(node->name, FLAGS_TAG) == 0) {
- profile->mFlags = parseOutputFlagNames((char *)node->value);
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- profile->loadGains(node);
- }
- node = node->next;
- }
- ALOGW_IF(profile->mSupportedDevices.isEmpty(),
- "loadOutput() invalid supported devices");
- ALOGW_IF(profile->mChannelMasks.size() == 0,
- "loadOutput() invalid supported channel masks");
- ALOGW_IF(profile->mSamplingRates.size() == 0,
- "loadOutput() invalid supported sampling rates");
- ALOGW_IF(profile->mFormats.size() == 0,
- "loadOutput() invalid supported formats");
- if (!profile->mSupportedDevices.isEmpty() &&
- (profile->mChannelMasks.size() != 0) &&
- (profile->mSamplingRates.size() != 0) &&
- (profile->mFormats.size() != 0)) {
-
- ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
- profile->mSupportedDevices.types(), profile->mFlags);
-
- mOutputProfiles.add(profile);
- return NO_ERROR;
- } else {
- return BAD_VALUE;
- }
-}
-
-status_t AudioPolicyManager::HwModule::loadDevice(cnode *root)
-{
- cnode *node = root->first_child;
-
- audio_devices_t type = AUDIO_DEVICE_NONE;
- while (node) {
- if (strcmp(node->name, DEVICE_TYPE) == 0) {
- type = parseDeviceNames((char *)node->value);
- break;
- }
- node = node->next;
- }
- if (type == AUDIO_DEVICE_NONE ||
- (!audio_is_input_device(type) && !audio_is_output_device(type))) {
- ALOGW("loadDevice() bad type %08x", type);
- return BAD_VALUE;
- }
- sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
- deviceDesc->mModule = this;
-
- node = root->first_child;
- while (node) {
- if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
- deviceDesc->mAddress = String8((char *)node->value);
- } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
- if (audio_is_input_device(type)) {
- deviceDesc->loadInChannels((char *)node->value);
- } else {
- deviceDesc->loadOutChannels((char *)node->value);
- }
- } else if (strcmp(node->name, GAINS_TAG) == 0) {
- deviceDesc->loadGains(node);
- }
- node = node->next;
- }
-
- ALOGV("loadDevice() adding device name %s type %08x address %s",
- deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
-
- mDeclaredDevices.add(deviceDesc);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addOutputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- mOutputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeOutputProfile(String8 name)
-{
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- if (mOutputProfiles[i]->mName == name) {
- mOutputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::addInputProfile(String8 name, const audio_config_t *config,
- audio_devices_t device, String8 address)
-{
- sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
-
- profile->mSamplingRates.add(config->sample_rate);
- profile->mChannelMasks.add(config->channel_mask);
- profile->mFormats.add(config->format);
-
- sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
- devDesc->mAddress = address;
- profile->mSupportedDevices.add(devDesc);
-
- ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
-
- mInputProfiles.add(profile);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::HwModule::removeInputProfile(String8 name)
-{
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- if (mInputProfiles[i]->mName == name) {
- mInputProfiles.removeAt(i);
- break;
- }
- }
-
- return NO_ERROR;
-}
-
-
-void AudioPolicyManager::HwModule::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, " - name: %s\n", mName);
- result.append(buffer);
- snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
- result.append(buffer);
- write(fd, result.string(), result.size());
- if (mOutputProfiles.size()) {
- write(fd, " - outputs:\n", strlen(" - outputs:\n"));
- for (size_t i = 0; i < mOutputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " output %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mOutputProfiles[i]->dump(fd);
- }
- }
- if (mInputProfiles.size()) {
- write(fd, " - inputs:\n", strlen(" - inputs:\n"));
- for (size_t i = 0; i < mInputProfiles.size(); i++) {
- snprintf(buffer, SIZE, " input %zu:\n", i);
- write(fd, buffer, strlen(buffer));
- mInputProfiles[i]->dump(fd);
- }
- }
- if (mDeclaredDevices.size()) {
- write(fd, " - devices:\n", strlen(" - devices:\n"));
- for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
- mDeclaredDevices[i]->dump(fd, 4, i);
- }
- }
-}
-
-// --- AudioPort class implementation
-
-
-AudioPolicyManager::AudioPort::AudioPort(const String8& name, audio_port_type_t type,
- audio_port_role_t role, const sp<HwModule>& module) :
- mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
-{
- mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
- ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
-}
-
-void AudioPolicyManager::AudioPort::attach(const sp<HwModule>& module) {
- mId = AudioPolicyManager::nextUniqueId();
- mModule = module;
-}
-
-void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const
-{
- port->role = mRole;
- port->type = mType;
- strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
- unsigned int i;
- for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
- if (mSamplingRates[i] != 0) {
- port->sample_rates[i] = mSamplingRates[i];
- }
- }
- port->num_sample_rates = i;
- for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
- if (mChannelMasks[i] != 0) {
- port->channel_masks[i] = mChannelMasks[i];
- }
- }
- port->num_channel_masks = i;
- for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
- if (mFormats[i] != 0) {
- port->formats[i] = mFormats[i];
- }
- }
- port->num_formats = i;
-
- ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
- for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
- port->gains[i] = mGains[i]->mGain;
- }
- port->num_gains = i;
-}
-
-void AudioPolicyManager::AudioPort::importAudioPort(const sp<AudioPort> port) {
- for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
- const uint32_t rate = port->mSamplingRates.itemAt(k);
- if (rate != 0) { // skip "dynamic" rates
- bool hasRate = false;
- for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
- if (rate == mSamplingRates.itemAt(l)) {
- hasRate = true;
- break;
- }
- }
- if (!hasRate) { // never import a sampling rate twice
- mSamplingRates.add(rate);
- }
- }
- }
- for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
- const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
- if (mask != 0) { // skip "dynamic" masks
- bool hasMask = false;
- for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
- if (mask == mChannelMasks.itemAt(l)) {
- hasMask = true;
- break;
- }
- }
- if (!hasMask) { // never import a channel mask twice
- mChannelMasks.add(mask);
- }
- }
- }
- for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
- const audio_format_t format = port->mFormats.itemAt(k);
- if (format != 0) { // skip "dynamic" formats
- bool hasFormat = false;
- for (size_t l = 0 ; l < mFormats.size() ; l++) {
- if (format == mFormats.itemAt(l)) {
- hasFormat = true;
- break;
- }
- }
- if (!hasFormat) { // never import a channel mask twice
- mFormats.add(format);
- }
- }
- }
- for (size_t k = 0 ; k < port->mGains.size() ; k++) {
- sp<AudioGain> gain = port->mGains.itemAt(k);
- if (gain != 0) {
- bool hasGain = false;
- for (size_t l = 0 ; l < mGains.size() ; l++) {
- if (gain == mGains.itemAt(l)) {
- hasGain = true;
- break;
- }
- }
- if (!hasGain) { // never import a gain twice
- mGains.add(gain);
- }
- }
- }
-}
-
-void AudioPolicyManager::AudioPort::clearCapabilities() {
- mChannelMasks.clear();
- mFormats.clear();
- mSamplingRates.clear();
- mGains.clear();
-}
-
-void AudioPolicyManager::AudioPort::loadSamplingRates(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
- // rates should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mSamplingRates.add(0);
- return;
- }
-
- while (str != NULL) {
- uint32_t rate = atoi(str);
- if (rate != 0) {
- ALOGV("loadSamplingRates() adding rate %d", rate);
- mSamplingRates.add(rate);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadFormats(char *name)
-{
- char *str = strtok(name, "|");
-
- // by convention, "0' in the first entry in mFormats indicates the supported formats
- // should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mFormats.add(AUDIO_FORMAT_DEFAULT);
- return;
- }
-
- while (str != NULL) {
- audio_format_t format = (audio_format_t)stringToEnum(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- str);
- if (format != AUDIO_FORMAT_DEFAULT) {
- mFormats.add(format);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadInChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadInChannels() %s", name);
-
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- ALOGV("loadInChannels() adding channelMask %04x", channelMask);
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
-}
-
-void AudioPolicyManager::AudioPort::loadOutChannels(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadOutChannels() %s", name);
-
- // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
- // masks should be read from the output stream after it is opened for the first time
- if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
- mChannelMasks.add(0);
- return;
- }
-
- while (str != NULL) {
- audio_channel_mask_t channelMask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- str);
- if (channelMask != 0) {
- mChannelMasks.add(channelMask);
- }
- str = strtok(NULL, "|");
- }
- return;
-}
-
-audio_gain_mode_t AudioPolicyManager::AudioPort::loadGainMode(char *name)
-{
- const char *str = strtok(name, "|");
-
- ALOGV("loadGainMode() %s", name);
- audio_gain_mode_t mode = 0;
- while (str != NULL) {
- mode |= (audio_gain_mode_t)stringToEnum(sGainModeNameToEnumTable,
- ARRAY_SIZE(sGainModeNameToEnumTable),
- str);
- str = strtok(NULL, "|");
- }
- return mode;
-}
-
-void AudioPolicyManager::AudioPort::loadGain(cnode *root, int index)
-{
- cnode *node = root->first_child;
-
- sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
-
- while (node) {
- if (strcmp(node->name, GAIN_MODE) == 0) {
- gain->mGain.mode = loadGainMode((char *)node->value);
- } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
- if (mUseInChannelMask) {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sInChannelsNameToEnumTable,
- ARRAY_SIZE(sInChannelsNameToEnumTable),
- (char *)node->value);
- } else {
- gain->mGain.channel_mask =
- (audio_channel_mask_t)stringToEnum(sOutChannelsNameToEnumTable,
- ARRAY_SIZE(sOutChannelsNameToEnumTable),
- (char *)node->value);
- }
- } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
- gain->mGain.min_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
- gain->mGain.max_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
- gain->mGain.default_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
- gain->mGain.step_value = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
- gain->mGain.min_ramp_ms = atoi((char *)node->value);
- } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
- gain->mGain.max_ramp_ms = atoi((char *)node->value);
- }
- node = node->next;
- }
-
- ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
- gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
-
- if (gain->mGain.mode == 0) {
- return;
- }
- mGains.add(gain);
-}
-
-void AudioPolicyManager::AudioPort::loadGains(cnode *root)
-{
- cnode *node = root->first_child;
- int index = 0;
- while (node) {
- ALOGV("loadGains() loading gain %s", node->name);
- loadGain(node, index++);
- node = node->next;
- }
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if (mSamplingRates[i] == samplingRate) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
- uint32_t *updatedSamplingRate) const
-{
- if (mSamplingRates.isEmpty()) {
- return NO_ERROR;
- }
-
- // Search for the closest supported sampling rate that is above (preferred)
- // or below (acceptable) the desired sampling rate, within a permitted ratio.
- // The sampling rates do not need to be sorted in ascending order.
- ssize_t maxBelow = -1;
- ssize_t minAbove = -1;
- uint32_t candidate;
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- candidate = mSamplingRates[i];
- if (candidate == samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- // candidate < desired
- if (candidate < samplingRate) {
- if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
- maxBelow = i;
- }
- // candidate > desired
- } else {
- if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
- minAbove = i;
- }
- }
- }
- // This uses hard-coded knowledge about AudioFlinger resampling ratios.
- // TODO Move these assumptions out.
- static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
- static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
- // due to approximation by an int32_t of the
- // phase increments
- // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
- if (minAbove >= 0) {
- candidate = mSamplingRates[minAbove];
- if (candidate / kMaxDownSampleRatio <= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // But if we have to up-sample from a lower sampling rate, that's OK.
- if (maxBelow >= 0) {
- candidate = mSamplingRates[maxBelow];
- if (candidate * kMaxUpSampleRatio >= samplingRate) {
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = candidate;
- }
- return NO_ERROR;
- }
- }
- // leave updatedSamplingRate unmodified
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- if (mChannelMasks[i] == channelMask) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
- const
-{
- if (mChannelMasks.isEmpty()) {
- return NO_ERROR;
- }
-
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- // FIXME Does not handle multi-channel automatic conversions yet
- audio_channel_mask_t supported = mChannelMasks[i];
- if (supported == channelMask) {
- return NO_ERROR;
- }
- if (isRecordThread) {
- // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
- // FIXME Abstract this out to a table.
- if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
- && channelMask == AUDIO_CHANNEL_IN_MONO) ||
- (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
- || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
- return NO_ERROR;
- }
- }
- }
- return BAD_VALUE;
-}
-
-status_t AudioPolicyManager::AudioPort::checkFormat(audio_format_t format) const
-{
- if (mFormats.isEmpty()) {
- return NO_ERROR;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if (mFormats[i] == format) {
- return NO_ERROR;
- }
- }
- return BAD_VALUE;
-}
-
-
-uint32_t AudioPolicyManager::AudioPort::pickSamplingRate() const
-{
- // special case for uninitialized dynamic profile
- if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
- return 0;
- }
-
- // For direct outputs, pick minimum sampling rate: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t samplingRate = UINT_MAX;
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return (samplingRate == UINT_MAX) ? 0 : samplingRate;
- }
-
- uint32_t samplingRate = 0;
- uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
-
- // For mixed output and inputs, use max mixer sampling rates. Do not
- // limit sampling rate otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxRate = UINT_MAX;
- }
- for (size_t i = 0; i < mSamplingRates.size(); i ++) {
- if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
- samplingRate = mSamplingRates[i];
- }
- }
- return samplingRate;
-}
-
-audio_channel_mask_t AudioPolicyManager::AudioPort::pickChannelMask() const
-{
- // special case for uninitialized dynamic profile
- if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
- return AUDIO_CHANNEL_NONE;
- }
- audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
-
- // For direct outputs, pick minimum channel count: this helps ensuring that the
- // channel count / sampling rate combination chosen will be supported by the connected
- // sink
- if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
- uint32_t channelCount = UINT_MAX;
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount < channelCount) && (cnlCount > 0)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
- }
-
- uint32_t channelCount = 0;
- uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
-
- // For mixed output and inputs, use max mixer channel count. Do not
- // limit channel count otherwise
- if (mType != AUDIO_PORT_TYPE_MIX) {
- maxCount = UINT_MAX;
- }
- for (size_t i = 0; i < mChannelMasks.size(); i ++) {
- uint32_t cnlCount;
- if (mUseInChannelMask) {
- cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
- } else {
- cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
- }
- if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
- channelMask = mChannelMasks[i];
- channelCount = cnlCount;
- }
- }
- return channelMask;
-}
-
-/* format in order of increasing preference */
-const audio_format_t AudioPolicyManager::AudioPort::sPcmFormatCompareTable[] = {
- AUDIO_FORMAT_DEFAULT,
- AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_FORMAT_PCM_8_24_BIT,
- AUDIO_FORMAT_PCM_24_BIT_PACKED,
- AUDIO_FORMAT_PCM_32_BIT,
- AUDIO_FORMAT_PCM_FLOAT,
-};
-
-int AudioPolicyManager::AudioPort::compareFormats(audio_format_t format1,
- audio_format_t format2)
-{
- // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
- // compressed format and better than any PCM format. This is by design of pickFormat()
- if (!audio_is_linear_pcm(format1)) {
- if (!audio_is_linear_pcm(format2)) {
- return 0;
- }
- return 1;
- }
- if (!audio_is_linear_pcm(format2)) {
- return -1;
- }
-
- int index1 = -1, index2 = -1;
- for (size_t i = 0;
- (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
- i ++) {
- if (sPcmFormatCompareTable[i] == format1) {
- index1 = i;
- }
- if (sPcmFormatCompareTable[i] == format2) {
- index2 = i;
- }
- }
- // format1 not found => index1 < 0 => format2 > format1
- // format2 not found => index2 < 0 => format2 < format1
- return index1 - index2;
-}
-
-audio_format_t AudioPolicyManager::AudioPort::pickFormat() const
-{
- // special case for uninitialized dynamic profile
- if (mFormats.size() == 1 && mFormats[0] == 0) {
- return AUDIO_FORMAT_DEFAULT;
- }
-
- audio_format_t format = AUDIO_FORMAT_DEFAULT;
- audio_format_t bestFormat =
- AudioPolicyManager::AudioPort::sPcmFormatCompareTable[
- ARRAY_SIZE(AudioPolicyManager::AudioPort::sPcmFormatCompareTable) - 1];
- // For mixed output and inputs, use best mixer output format. Do not
- // limit format otherwise
- if ((mType != AUDIO_PORT_TYPE_MIX) ||
- ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
- (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
- bestFormat = AUDIO_FORMAT_INVALID;
- }
-
- for (size_t i = 0; i < mFormats.size(); i ++) {
- if ((compareFormats(mFormats[i], format) > 0) &&
- (compareFormats(mFormats[i], bestFormat) <= 0)) {
- format = mFormats[i];
- }
- }
- return format;
-}
-
-status_t AudioPolicyManager::AudioPort::checkGain(const struct audio_gain_config *gainConfig,
- int index) const
-{
- if (index < 0 || (size_t)index >= mGains.size()) {
- return BAD_VALUE;
- }
- return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPolicyManager::AudioPort::dump(int fd, int spaces) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- if (mName.size() != 0) {
- snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
- result.append(buffer);
- }
-
- if (mSamplingRates.size() != 0) {
- snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- if (i == 0 && mSamplingRates[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
- }
- result.append(buffer);
- result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mChannelMasks.size() != 0) {
- snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
-
- if (i == 0 && mChannelMasks[i] == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
- }
- result.append(buffer);
- result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
-
- if (mFormats.size() != 0) {
- snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
- result.append(buffer);
- for (size_t i = 0; i < mFormats.size(); i++) {
- const char *formatStr = enumToString(sFormatNameToEnumTable,
- ARRAY_SIZE(sFormatNameToEnumTable),
- mFormats[i]);
- if (i == 0 && strcmp(formatStr, "") == 0) {
- snprintf(buffer, SIZE, "Dynamic");
- } else {
- snprintf(buffer, SIZE, "%s", formatStr);
- }
- result.append(buffer);
- result.append(i == (mFormats.size() - 1) ? "" : ", ");
- }
- result.append("\n");
- }
- write(fd, result.string(), result.size());
- if (mGains.size() != 0) {
- snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
- write(fd, buffer, strlen(buffer) + 1);
- result.append(buffer);
- for (size_t i = 0; i < mGains.size(); i++) {
- mGains[i]->dump(fd, spaces + 2, i);
- }
- }
-}
-
-// --- AudioGain class implementation
-
-AudioPolicyManager::AudioGain::AudioGain(int index, bool useInChannelMask)
-{
- mIndex = index;
- mUseInChannelMask = useInChannelMask;
- memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioPolicyManager::AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
- config->index = mIndex;
- config->mode = mGain.mode;
- config->channel_mask = mGain.channel_mask;
- if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- config->values[0] = mGain.default_value;
- } else {
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- config->values[i] = mGain.default_value;
- }
- }
- if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- config->ramp_duration_ms = mGain.min_ramp_ms;
- }
-}
-
-status_t AudioPolicyManager::AudioGain::checkConfig(const struct audio_gain_config *config)
-{
- if ((config->mode & ~mGain.mode) != 0) {
- return BAD_VALUE;
- }
- if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
- if ((config->values[0] < mGain.min_value) ||
- (config->values[0] > mGain.max_value)) {
- return BAD_VALUE;
- }
- } else {
- if ((config->channel_mask & ~mGain.channel_mask) != 0) {
- return BAD_VALUE;
- }
- uint32_t numValues;
- if (mUseInChannelMask) {
- numValues = audio_channel_count_from_in_mask(config->channel_mask);
- } else {
- numValues = audio_channel_count_from_out_mask(config->channel_mask);
- }
- for (size_t i = 0; i < numValues; i++) {
- if ((config->values[i] < mGain.min_value) ||
- (config->values[i] > mGain.max_value)) {
- return BAD_VALUE;
- }
- }
- }
- if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
- if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
- (config->ramp_duration_ms > mGain.max_ramp_ms)) {
- return BAD_VALUE;
- }
- }
- return NO_ERROR;
-}
-
-void AudioPolicyManager::AudioGain::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
- result.append(buffer);
-
- write(fd, result.string(), result.size());
-}
-
-// --- AudioPortConfig class implementation
-
-AudioPolicyManager::AudioPortConfig::AudioPortConfig()
-{
- mSamplingRate = 0;
- mChannelMask = AUDIO_CHANNEL_NONE;
- mFormat = AUDIO_FORMAT_INVALID;
- mGain.index = -1;
-}
-
-status_t AudioPolicyManager::AudioPortConfig::applyAudioPortConfig(
- const struct audio_port_config *config,
- struct audio_port_config *backupConfig)
-{
- struct audio_port_config localBackupConfig;
- status_t status = NO_ERROR;
-
- localBackupConfig.config_mask = config->config_mask;
- toAudioPortConfig(&localBackupConfig);
-
- sp<AudioPort> audioport = getAudioPort();
- if (audioport == 0) {
- status = NO_INIT;
- goto exit;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- status = audioport->checkExactSamplingRate(config->sample_rate);
- if (status != NO_ERROR) {
- goto exit;
- }
- mSamplingRate = config->sample_rate;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- status = audioport->checkExactChannelMask(config->channel_mask);
- if (status != NO_ERROR) {
- goto exit;
- }
- mChannelMask = config->channel_mask;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- status = audioport->checkFormat(config->format);
- if (status != NO_ERROR) {
- goto exit;
- }
- mFormat = config->format;
- }
- if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- status = audioport->checkGain(&config->gain, config->gain.index);
- if (status != NO_ERROR) {
- goto exit;
- }
- mGain = config->gain;
- }
-
-exit:
- if (status != NO_ERROR) {
- applyAudioPortConfig(&localBackupConfig);
- }
- if (backupConfig != NULL) {
- *backupConfig = localBackupConfig;
- }
- return status;
-}
-
-void AudioPolicyManager::AudioPortConfig::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
- dstConfig->sample_rate = mSamplingRate;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
- dstConfig->sample_rate = srcConfig->sample_rate;
- }
- } else {
- dstConfig->sample_rate = 0;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
- dstConfig->channel_mask = mChannelMask;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
- dstConfig->channel_mask = srcConfig->channel_mask;
- }
- } else {
- dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
- dstConfig->format = mFormat;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
- dstConfig->format = srcConfig->format;
- }
- } else {
- dstConfig->format = AUDIO_FORMAT_INVALID;
- }
- if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
- dstConfig->gain = mGain;
- if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
- dstConfig->gain = srcConfig->gain;
- }
- } else {
- dstConfig->gain.index = -1;
- }
- if (dstConfig->gain.index != -1) {
- dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
- } else {
- dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
- }
-}
-
-// --- IOProfile class implementation
-
-AudioPolicyManager::IOProfile::IOProfile(const String8& name, audio_port_role_t role,
- const sp<HwModule>& module)
- : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
-{
-}
-
-AudioPolicyManager::IOProfile::~IOProfile()
-{
-}
-
-// checks if the IO profile is compatible with specified parameters.
-// Sampling rate, format and channel mask must be specified in order to
-// get a valid a match
-bool AudioPolicyManager::IOProfile::isCompatibleProfile(audio_devices_t device,
- String8 address,
- uint32_t samplingRate,
- uint32_t *updatedSamplingRate,
- audio_format_t format,
- audio_channel_mask_t channelMask,
- uint32_t flags) const
-{
- const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
- const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
- ALOG_ASSERT(isPlaybackThread != isRecordThread);
-
- if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
- return false;
- }
-
- if (samplingRate == 0) {
- return false;
- }
- uint32_t myUpdatedSamplingRate = samplingRate;
- if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
- return false;
- }
- if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
- NO_ERROR) {
- return false;
- }
-
- if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
- return false;
- }
-
- if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
- checkExactChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
- if (isRecordThread && (!audio_is_input_channel(channelMask) ||
- checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
- return false;
- }
-
- if (isPlaybackThread && (mFlags & flags) != flags) {
- return false;
- }
- // The only input flag that is allowed to be different is the fast flag.
- // An existing fast stream is compatible with a normal track request.
- // An existing normal stream is compatible with a fast track request,
- // but the fast request will be denied by AudioFlinger and converted to normal track.
- if (isRecordThread && ((mFlags ^ flags) &
- ~AUDIO_INPUT_FLAG_FAST)) {
- return false;
- }
-
- if (updatedSamplingRate != NULL) {
- *updatedSamplingRate = myUpdatedSamplingRate;
- }
- return true;
-}
-
-void AudioPolicyManager::IOProfile::dump(int fd)
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- AudioPort::dump(fd, 4);
-
- snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
- result.append(buffer);
- snprintf(buffer, SIZE, " - devices:\n");
- result.append(buffer);
- write(fd, result.string(), result.size());
- for (size_t i = 0; i < mSupportedDevices.size(); i++) {
- mSupportedDevices[i]->dump(fd, 6, i);
- }
-}
-
-void AudioPolicyManager::IOProfile::log()
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- ALOGV(" - sampling rates: ");
- for (size_t i = 0; i < mSamplingRates.size(); i++) {
- ALOGV(" %d", mSamplingRates[i]);
- }
-
- ALOGV(" - channel masks: ");
- for (size_t i = 0; i < mChannelMasks.size(); i++) {
- ALOGV(" 0x%04x", mChannelMasks[i]);
- }
-
- ALOGV(" - formats: ");
- for (size_t i = 0; i < mFormats.size(); i++) {
- ALOGV(" 0x%08x", mFormats[i]);
- }
-
- ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
- ALOGV(" - flags: 0x%04x\n", mFlags);
-}
-
-
-// --- DeviceDescriptor implementation
-
-String8 AudioPolicyManager::DeviceDescriptor::emptyNameStr = String8("");
-
-AudioPolicyManager::DeviceDescriptor::DeviceDescriptor(const 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("")
-{
-}
-
-bool AudioPolicyManager::DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
-{
- // Devices are considered equal if they:
- // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
- // - have the same address or one device does not specify the address
- // - have the same channel mask or one device does not specify the channel mask
- return (mDeviceType == other->mDeviceType) &&
- (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
- (mChannelMask == 0 || other->mChannelMask == 0 ||
- mChannelMask == other->mChannelMask);
-}
-
-void AudioPolicyManager::DeviceDescriptor::loadGains(cnode *root)
-{
- AudioPort::loadGains(root);
- if (mGains.size() > 0) {
- mGains[0]->getDefaultConfig(&mGain);
- }
-}
-
-
-void AudioPolicyManager::DeviceVector::refreshTypes()
-{
- mDeviceTypes = AUDIO_DEVICE_NONE;
- for(size_t i = 0; i < size(); i++) {
- mDeviceTypes |= itemAt(i)->mDeviceType;
- }
- ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
-}
-
-ssize_t AudioPolicyManager::DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
-{
- for(size_t i = 0; i < size(); i++) {
- if (item->equals(itemAt(i))) {
- return i;
- }
- }
- return -1;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::add(const sp<DeviceDescriptor>& item)
-{
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ret = SortedVector::add(item);
- if (ret >= 0) {
- refreshTypes();
- }
- } else {
- ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
- ret = -1;
- }
- return ret;
-}
-
-ssize_t AudioPolicyManager::DeviceVector::remove(const sp<DeviceDescriptor>& item)
-{
- size_t i;
- ssize_t ret = indexOf(item);
-
- if (ret < 0) {
- ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
- } else {
- ret = SortedVector::removeAt(ret);
- if (ret >= 0) {
- refreshTypes();
- }
- }
- return ret;
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromType(audio_devices_t types)
-{
- DeviceVector deviceList;
-
- uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
- types &= ~role_bit;
-
- while (types) {
- uint32_t i = 31 - __builtin_clz(types);
- uint32_t type = 1 << i;
- types &= ~type;
- add(new DeviceDescriptor(String8("device_type"), type | role_bit));
- }
-}
-
-void AudioPolicyManager::DeviceVector::loadDevicesFromName(char *name,
- const DeviceVector& declaredDevices)
-{
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- audio_devices_t type = stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- if (type != AUDIO_DEVICE_NONE) {
- sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
- if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
- type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
- dev->mAddress = String8("0");
- }
- add(dev);
- } else {
- sp<DeviceDescriptor> deviceDesc =
- declaredDevices.getDeviceFromName(String8(devName));
- if (deviceDesc != 0) {
- add(deviceDesc);
- }
- }
- }
- devName = strtok(NULL, "|");
- }
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDevice(
- audio_devices_t type, String8 address) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (address == "" || itemAt(i)->mAddress == address) {
- device = itemAt(i);
- if (itemAt(i)->mAddress == address) {
- break;
- }
- }
- }
- }
- ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
- type, address.string(), device.get());
- return device;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromId(
- audio_port_handle_t id) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->getHandle() == id) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType(
- audio_devices_t type) const
-{
- DeviceVector devices;
- for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
- if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
- devices.add(itemAt(i));
- type &= ~itemAt(i)->mDeviceType;
- ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
- itemAt(i)->mDeviceType, itemAt(i).get());
- }
- }
- return devices;
-}
-
-AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromTypeAddr(
- audio_devices_t type, String8 address) const
-{
- DeviceVector devices;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mDeviceType == type) {
- if (itemAt(i)->mAddress == address) {
- devices.add(itemAt(i));
- }
- }
- }
- return devices;
-}
-
-sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::DeviceVector::getDeviceFromName(
- const String8& name) const
-{
- sp<DeviceDescriptor> device;
- for (size_t i = 0; i < size(); i++) {
- if (itemAt(i)->mName == name) {
- device = itemAt(i);
- break;
- }
- }
- return device;
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPortConfig(
- struct audio_port_config *dstConfig,
- const struct audio_port_config *srcConfig) const
-{
- dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
- if (srcConfig != NULL) {
- dstConfig->config_mask |= srcConfig->config_mask;
- }
-
- AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
- dstConfig->id = mId;
- dstConfig->role = audio_is_output_device(mDeviceType) ?
- AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
- dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
- dstConfig->ext.device.type = mDeviceType;
-
- //TODO Understand why this test is necessary. i.e. why at boot time does it crash
- // without the test?
- // This has been demonstrated to NOT be true (at start up)
- // ALOG_ASSERT(mModule != NULL);
- dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL;
- strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-void AudioPolicyManager::DeviceDescriptor::toAudioPort(struct audio_port *port) const
-{
- ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
- AudioPort::toAudioPort(port);
- port->id = mId;
- toAudioPortConfig(&port->active_config);
- port->ext.device.type = mDeviceType;
- port->ext.device.hw_module = mModule->mHandle;
- strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
-}
-
-status_t AudioPolicyManager::DeviceDescriptor::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
- result.append(buffer);
- if (mId != 0) {
- snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
- enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mDeviceType));
- result.append(buffer);
- if (mAddress.size() != 0) {
- snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
- result.append(buffer);
- }
- write(fd, result.string(), result.size());
- AudioPort::dump(fd, spaces);
-
- return NO_ERROR;
-}
-
-status_t AudioPolicyManager::AudioPatch::dump(int fd, int spaces, int index) const
-{
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
-
- snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
- result.append(buffer);
- snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sources; i++) {
- if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sources[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sources[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
- }
- result.append(buffer);
- }
- snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
- result.append(buffer);
- for (size_t i = 0; i < mPatch.num_sinks; i++) {
- if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
- snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
- mPatch.sinks[i].id, enumToString(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- mPatch.sinks[i].ext.device.type));
- } else {
- snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
- mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
- }
- result.append(buffer);
- }
-
- write(fd, result.string(), result.size());
- return NO_ERROR;
-}
// --- audio_policy.conf file parsing
-
-uint32_t AudioPolicyManager::parseOutputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sOutputFlagNameToEnumTable,
- ARRAY_SIZE(sOutputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- //force direct flag if offload flag is set: offloading implies a direct output stream
- // and all common behaviors are driven by checking only the direct flag
- // this should normally be set appropriately in the policy configuration file
- if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
- flag |= AUDIO_OUTPUT_FLAG_DIRECT;
- }
-
- return flag;
-}
-
-uint32_t AudioPolicyManager::parseInputFlagNames(char *name)
-{
- uint32_t flag = 0;
-
- // it is OK to cast name to non const here as we are not going to use it after
- // strtok() modifies it
- char *flagName = strtok(name, "|");
- while (flagName != NULL) {
- if (strlen(flagName) != 0) {
- flag |= stringToEnum(sInputFlagNameToEnumTable,
- ARRAY_SIZE(sInputFlagNameToEnumTable),
- flagName);
- }
- flagName = strtok(NULL, "|");
- }
- return flag;
-}
-
-audio_devices_t AudioPolicyManager::parseDeviceNames(char *name)
-{
- uint32_t device = 0;
-
- char *devName = strtok(name, "|");
- while (devName != NULL) {
- if (strlen(devName) != 0) {
- device |= stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- devName);
- }
- devName = strtok(NULL, "|");
- }
- return device;
-}
-
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadHwModule(cnode *root)
{
status_t status = NAME_NOT_FOUND;
@@ -7903,6 +5557,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadHwModules(cnode *root)
{
cnode *node = config_find(root, AUDIO_HW_MODULE_TAG);
@@ -7918,6 +5573,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module)
{
cnode *node = config_find(root, GLOBAL_CONFIG_TAG);
@@ -7938,9 +5594,10 @@
ALOGV("loadGlobalConfig() Attached Output Devices %08x",
mAvailableOutputDevices.types());
} else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) {
- audio_devices_t device = (audio_devices_t)stringToEnum(sDeviceNameToEnumTable,
- ARRAY_SIZE(sDeviceNameToEnumTable),
- (char *)node->value);
+ audio_devices_t device = (audio_devices_t)ConfigParsingUtils::stringToEnum(
+ sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ (char *)node->value);
if (device != AUDIO_DEVICE_NONE) {
mDefaultOutputDevice = new DeviceDescriptor(String8("default-output"), device);
} else {
@@ -7952,7 +5609,7 @@
declaredDevices);
ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types());
} else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) {
- mSpeakerDrcEnabled = stringToBool((char *)node->value);
+ mSpeakerDrcEnabled = ConfigParsingUtils::stringToBool((char *)node->value);
ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled);
} else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) {
uint32_t major, minor;
@@ -7965,6 +5622,7 @@
}
}
+// TODO candidate to be moved to ConfigParsingUtils
status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path)
{
cnode *root;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
new file mode 100644
index 0000000..61ea6f2
--- /dev/null
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <cutils/config_utils.h>
+#include <cutils/misc.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/SortedVector.h>
+#include <media/AudioPolicy.h>
+#include "AudioPolicyInterface.h"
+
+#include "Gains.h"
+#include "Ports.h"
+#include "ConfigParsingUtils.h"
+#include "Devices.h"
+#include "IOProfile.h"
+#include "HwModule.h"
+#include "AudioInputDescriptor.h"
+#include "AudioOutputDescriptor.h"
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// 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
+
+#define MAX_MIXER_SAMPLING_RATE 48000
+#define MAX_MIXER_CHANNEL_COUNT 8
+
+// ----------------------------------------------------------------------------
+// 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);
+ virtual ~AudioPolicyManager();
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ 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,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ virtual status_t getOutputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *output,
+ audio_session_t session,
+ audio_stream_type_t *stream,
+ uint32_t samplingRate,
+ 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,
+ audio_session_t session);
+ virtual status_t stopOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual void releaseOutput(audio_io_handle_t output,
+ audio_stream_type_t stream,
+ audio_session_t session);
+ virtual status_t getInputForAttr(const audio_attributes_t *attr,
+ audio_io_handle_t *input,
+ audio_session_t session,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags,
+ input_type_t *inputType);
+
+ // indicates to the audio policy manager that the input starts being used.
+ virtual status_t startInput(audio_io_handle_t input,
+ audio_session_t session);
+
+ // indicates to the audio policy manager that the input stops being used.
+ virtual status_t stopInput(audio_io_handle_t input,
+ audio_session_t session);
+ virtual void releaseInput(audio_io_handle_t input,
+ audio_session_t session);
+ 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 strategy corresponding to the given audio attributes
+ virtual uint32_t getStrategyForAttr(const audio_attributes_t *attr);
+
+ // 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
+ // For the base implementation, "remotely" means playing during screen mirroring which
+ // uses an output for playback with a non-empty, non "0" address.
+ 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 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);
+
+ virtual status_t acquireSoundTriggerSession(audio_session_t *session,
+ audio_io_handle_t *ioHandle,
+ audio_devices_t *device);
+
+ virtual status_t releaseSoundTriggerSession(audio_session_t session);
+
+ virtual status_t registerPolicyMixes(Vector<AudioMix> mixes);
+ virtual status_t unregisterPolicyMixes(Vector<AudioMix> mixes);
+
+ // Audio policy configuration file parsing (audio_policy.conf)
+ // TODO candidates to be moved to ConfigParsingUtils
+ void loadHwModule(cnode *root);
+ void loadHwModules(cnode *root);
+ void loadGlobalConfig(cnode *root, const sp<HwModule>& module);
+ status_t loadAudioPolicyConfig(const char *path);
+ void defaultAudioPolicyConfig(void);
+
+ // return the strategy corresponding to a given stream type
+ static routing_strategy getStrategy(audio_stream_type_t stream);
+
+ static uint32_t nextUniqueId();
+protected:
+
+ class EffectDescriptor : public RefBase
+ {
+ 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, sp<AudioOutputDescriptor> outputDesc);
+ void addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[],
+ // otherwise it is determine by current state
+ // (device connected,phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDevicesAndOutputs() is called.
+ virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy,
+ bool fromCache);
+
+ // change the route of the specified output. Returns the number of ms we have slept to
+ // allow new routing to take effect in certain cases.
+ virtual uint32_t setOutputDevice(audio_io_handle_t output,
+ audio_devices_t device,
+ bool force = false,
+ int delayMs = 0,
+ audio_patch_handle_t *patchHandle = NULL,
+ const char* address = 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(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);
+
+ uint32_t activeInputsCount() const;
+
+ // 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(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
+ virtual 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(const sp<DeviceDescriptor> devDesc,
+ 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);
+
+ // close an input.
+ void closeInput(audio_io_handle_t input);
+
+ // 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(const sp<EffectDescriptor>& effectDesc, bool enabled);
+
+ SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs);
+ bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
+ SortedVector<audio_io_handle_t>& outputs2);
+
+ // mute/unmute strategies using an incompatible device combination
+ // if muting, wait for the audio in pcm buffer to be drained before proceeding
+ // if unmuting, unmute only after the specified delay
+ // Returns the number of ms waited
+ virtual uint32_t checkDeviceMuteStrategies(sp<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,
+ audio_format_t format);
+ // samplingRate parameter is an in/out and so may be modified
+ sp<IOProfile> getInputProfile(audio_devices_t device,
+ String8 address,
+ uint32_t& samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_input_flags_t flags);
+ 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();
+
+ virtual status_t addAudioPatch(audio_patch_handle_t handle,
+ const sp<AudioPatch>& patch);
+ virtual status_t removeAudioPatch(audio_patch_handle_t handle);
+
+ sp<AudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const;
+ sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const;
+ sp<HwModule> getModuleForDevice(audio_devices_t device) const;
+ sp<HwModule> getModuleFromName(const char *name) const;
+ audio_devices_t availablePrimaryOutputDevices();
+ audio_devices_t availablePrimaryInputDevices();
+
+ void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0);
+
+
+ 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, sp<AudioOutputDescriptor> > mOutputs;
+ // copy of mOutputs before setDeviceConnectionState() opens new outputs
+ // reset to mOutputs when updateDevicesAndOutputs() is called.
+ DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > mPreviousOutputs;
+ DefaultKeyedVector<audio_io_handle_t, sp<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, sp<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 < sp<HwModule> > mHwModules;
+ static volatile int32_t mNextUniqueId;
+ volatile int32_t mAudioPortGeneration;
+
+ DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > mAudioPatches;
+
+ DefaultKeyedVector<audio_session_t, audio_io_handle_t> mSoundTriggerSessions;
+
+ sp<AudioPatch> mCallTxPatch;
+ sp<AudioPatch> mCallRxPatch;
+
+ // 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
+
+ // custom mix entry in mPolicyMixes
+ class AudioPolicyMix : public RefBase {
+ public:
+ AudioPolicyMix() {}
+
+ AudioMix mMix; // Audio policy mix descriptor
+ sp<AudioOutputDescriptor> mOutput; // Corresponding output stream
+ };
+ DefaultKeyedVector<String8, sp<AudioPolicyMix> > mPolicyMixes; // list of registered mixes
+
+
+#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
+
+ static bool isVirtualInputDevice(audio_devices_t device);
+
+ uint32_t nextAudioPortGeneration();
+private:
+ // updates device caching and output for streams that can influence the
+ // routing of notifications
+ void handleNotificationRoutingForStream(audio_stream_type_t stream);
+ static bool deviceDistinguishesOnAddress(audio_devices_t device);
+ // find the outputs on a given output descriptor that have the given address.
+ // to be called on an AudioOutputDescriptor whose supported devices (as defined
+ // in mProfile->mSupportedDevices) matches the device whose address is to be matched.
+ // see deviceDistinguishesOnAddress(audio_devices_t) for whether the device type is one
+ // where addresses are used to distinguish between one connected device and another.
+ void findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/,
+ const audio_devices_t device /*in*/,
+ const String8 address /*in*/,
+ SortedVector<audio_io_handle_t>& outputs /*out*/);
+ uint32_t curAudioPortGeneration() const { return mAudioPortGeneration; }
+ // internal method to return the output handle for the given device and format
+ audio_io_handle_t getOutputForDevice(
+ audio_devices_t device,
+ audio_session_t session,
+ audio_stream_type_t stream,
+ uint32_t samplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ audio_output_flags_t flags,
+ const audio_offload_info_t *offloadInfo);
+ // internal function to derive a stream type value from audio attributes
+ audio_stream_type_t streamTypefromAttributesInt(const audio_attributes_t *attr);
+ // return true if any output is playing anything besides the stream to ignore
+ bool isAnyOutputActive(audio_stream_type_t streamToIgnore);
+ // 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);
+
+ // select input device corresponding to requested audio source and return associated policy
+ // mix if any. Calls getDeviceForInputSource().
+ audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource,
+ AudioMix **policyMix = NULL);
+
+ // Called by setDeviceConnectionState().
+ status_t setDeviceConnectionStateInt(audio_devices_t device,
+ audio_policy_dev_state_t state,
+ const char *device_address,
+ const char *device_name);
+ sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
+ const char *device_address,
+ const char *device_name);
+};
+
+};
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
new file mode 100644
index 0000000..1afd487
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.cpp
@@ -0,0 +1,121 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::ConfigParsingUtils"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+//static
+uint32_t ConfigParsingUtils::stringToEnum(const struct StringToEnum *table,
+ size_t size,
+ const char *name)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (strcmp(table[i].name, name) == 0) {
+ ALOGV("stringToEnum() found %s", table[i].name);
+ return table[i].value;
+ }
+ }
+ return 0;
+}
+
+//static
+const char *ConfigParsingUtils::enumToString(const struct StringToEnum *table,
+ size_t size,
+ uint32_t value)
+{
+ for (size_t i = 0; i < size; i++) {
+ if (table[i].value == value) {
+ return table[i].name;
+ }
+ }
+ return "";
+}
+
+//static
+bool ConfigParsingUtils::stringToBool(const char *value)
+{
+ return ((strcasecmp("true", value) == 0) || (strcmp("1", value) == 0));
+}
+
+
+// --- audio_policy.conf file parsing
+//static
+uint32_t ConfigParsingUtils::parseOutputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= ConfigParsingUtils::stringToEnum(sOutputFlagNameToEnumTable,
+ ARRAY_SIZE(sOutputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ //force direct flag if offload flag is set: offloading implies a direct output stream
+ // and all common behaviors are driven by checking only the direct flag
+ // this should normally be set appropriately in the policy configuration file
+ if ((flag & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) {
+ flag |= AUDIO_OUTPUT_FLAG_DIRECT;
+ }
+
+ return flag;
+}
+
+//static
+uint32_t ConfigParsingUtils::parseInputFlagNames(char *name)
+{
+ uint32_t flag = 0;
+
+ // it is OK to cast name to non const here as we are not going to use it after
+ // strtok() modifies it
+ char *flagName = strtok(name, "|");
+ while (flagName != NULL) {
+ if (strlen(flagName) != 0) {
+ flag |= stringToEnum(sInputFlagNameToEnumTable,
+ ARRAY_SIZE(sInputFlagNameToEnumTable),
+ flagName);
+ }
+ flagName = strtok(NULL, "|");
+ }
+ return flag;
+}
+
+//static
+audio_devices_t ConfigParsingUtils::parseDeviceNames(char *name)
+{
+ uint32_t device = 0;
+
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ device |= stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ }
+ devName = strtok(NULL, "|");
+ }
+ return device;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/ConfigParsingUtils.h b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
new file mode 100644
index 0000000..7969661
--- /dev/null
+++ b/services/audiopolicy/managerdefault/ConfigParsingUtils.h
@@ -0,0 +1,159 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+// Definitions for audio_policy.conf file parsing
+// ----------------------------------------------------------------------------
+
+struct StringToEnum {
+ const char *name;
+ uint32_t value;
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+const StringToEnum sDeviceNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_EARPIECE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPEAKER_SAFE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_WIRED_HEADPHONE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_ALL_USB),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_TELEPHONY_TX),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_HDMI_ARC),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_FM),
+ STRING_TO_ENUM(AUDIO_DEVICE_OUT_AUX_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AMBIENT),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BUILTIN_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ALL_SCO),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_WIRED_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_AUX_DIGITAL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_HDMI),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TELEPHONY_RX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_VOICE_CALL),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BACK_MIC),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_REMOTE_SUBMIX),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_ACCESSORY),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_USB_DEVICE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_FM_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_TV_TUNER),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LINE),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_SPDIF),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_BLUETOOTH_A2DP),
+ STRING_TO_ENUM(AUDIO_DEVICE_IN_LOOPBACK),
+};
+
+const StringToEnum sOutputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DIRECT),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_PRIMARY),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_NON_BLOCKING),
+ STRING_TO_ENUM(AUDIO_OUTPUT_FLAG_HW_AV_SYNC),
+};
+
+const StringToEnum sInputFlagNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_FAST),
+ STRING_TO_ENUM(AUDIO_INPUT_FLAG_HW_HOTWORD),
+};
+
+const StringToEnum sFormatNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_32_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_8_24_BIT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_FLOAT),
+ STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_PACKED),
+ STRING_TO_ENUM(AUDIO_FORMAT_MP3),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_MAIN),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SSR),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LTP),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_SCALABLE),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ERLC),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_LD),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_HE_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_AAC_ELD),
+ STRING_TO_ENUM(AUDIO_FORMAT_VORBIS),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V1),
+ STRING_TO_ENUM(AUDIO_FORMAT_HE_AAC_V2),
+ STRING_TO_ENUM(AUDIO_FORMAT_OPUS),
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+};
+
+const StringToEnum sOutChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+const StringToEnum sInChannelsNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
+};
+
+const StringToEnum sGainModeNameToEnumTable[] = {
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_JOINT),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_CHANNELS),
+ STRING_TO_ENUM(AUDIO_GAIN_MODE_RAMP),
+};
+
+class ConfigParsingUtils
+{
+public:
+ 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 uint32_t parseOutputFlagNames(char *name);
+ static uint32_t parseInputFlagNames(char *name);
+ static audio_devices_t parseDeviceNames(char *name);
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.cpp b/services/audiopolicy/managerdefault/Devices.cpp
new file mode 100644
index 0000000..13c8bbc
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.cpp
@@ -0,0 +1,282 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Devices"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+String8 DeviceDescriptor::emptyNameStr = String8("");
+
+DeviceDescriptor::DeviceDescriptor(const 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("")
+{
+
+}
+
+bool DeviceDescriptor::equals(const sp<DeviceDescriptor>& other) const
+{
+ // Devices are considered equal if they:
+ // - are of the same type (a device type cannot be AUDIO_DEVICE_NONE)
+ // - have the same address or one device does not specify the address
+ // - have the same channel mask or one device does not specify the channel mask
+ return (mDeviceType == other->mDeviceType) &&
+ (mAddress == "" || other->mAddress == "" || mAddress == other->mAddress) &&
+ (mChannelMask == 0 || other->mChannelMask == 0 ||
+ mChannelMask == other->mChannelMask);
+}
+
+void DeviceDescriptor::loadGains(cnode *root)
+{
+ AudioPort::loadGains(root);
+ if (mGains.size() > 0) {
+ mGains[0]->getDefaultConfig(&mGain);
+ }
+}
+
+void DeviceVector::refreshTypes()
+{
+ mDeviceTypes = AUDIO_DEVICE_NONE;
+ for(size_t i = 0; i < size(); i++) {
+ mDeviceTypes |= itemAt(i)->mDeviceType;
+ }
+ ALOGV("DeviceVector::refreshTypes() mDeviceTypes %08x", mDeviceTypes);
+}
+
+ssize_t DeviceVector::indexOf(const sp<DeviceDescriptor>& item) const
+{
+ for(size_t i = 0; i < size(); i++) {
+ if (item->equals(itemAt(i))) {
+ return i;
+ }
+ }
+ return -1;
+}
+
+ssize_t DeviceVector::add(const sp<DeviceDescriptor>& item)
+{
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ret = SortedVector::add(item);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ } else {
+ ALOGW("DeviceVector::add device %08x already in", item->mDeviceType);
+ ret = -1;
+ }
+ return ret;
+}
+
+ssize_t DeviceVector::remove(const sp<DeviceDescriptor>& item)
+{
+ size_t i;
+ ssize_t ret = indexOf(item);
+
+ if (ret < 0) {
+ ALOGW("DeviceVector::remove device %08x not in", item->mDeviceType);
+ } else {
+ ret = SortedVector::removeAt(ret);
+ if (ret >= 0) {
+ refreshTypes();
+ }
+ }
+ return ret;
+}
+
+void DeviceVector::loadDevicesFromType(audio_devices_t types)
+{
+ DeviceVector deviceList;
+
+ uint32_t role_bit = AUDIO_DEVICE_BIT_IN & types;
+ types &= ~role_bit;
+
+ while (types) {
+ uint32_t i = 31 - __builtin_clz(types);
+ uint32_t type = 1 << i;
+ types &= ~type;
+ add(new DeviceDescriptor(String8("device_type"), type | role_bit));
+ }
+}
+
+void DeviceVector::loadDevicesFromName(char *name,
+ const DeviceVector& declaredDevices)
+{
+ char *devName = strtok(name, "|");
+ while (devName != NULL) {
+ if (strlen(devName) != 0) {
+ audio_devices_t type = ConfigParsingUtils::stringToEnum(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ devName);
+ if (type != AUDIO_DEVICE_NONE) {
+ sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(name), type);
+ if (type == AUDIO_DEVICE_IN_REMOTE_SUBMIX ||
+ type == AUDIO_DEVICE_OUT_REMOTE_SUBMIX ) {
+ dev->mAddress = String8("0");
+ }
+ add(dev);
+ } else {
+ sp<DeviceDescriptor> deviceDesc =
+ declaredDevices.getDeviceFromName(String8(devName));
+ if (deviceDesc != 0) {
+ add(deviceDesc);
+ }
+ }
+ }
+ devName = strtok(NULL, "|");
+ }
+}
+
+sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, String8 address) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (address == "" || itemAt(i)->mAddress == address) {
+ device = itemAt(i);
+ if (itemAt(i)->mAddress == address) {
+ break;
+ }
+ }
+ }
+ }
+ ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
+ type, address.string(), device.get());
+ return device;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromId(audio_port_handle_t id) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->getHandle() == id) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) {
+ if (itemAt(i)->mDeviceType & type & ~AUDIO_DEVICE_BIT_IN) {
+ devices.add(itemAt(i));
+ type &= ~itemAt(i)->mDeviceType;
+ ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
+ itemAt(i)->mDeviceType, itemAt(i).get());
+ }
+ }
+ return devices;
+}
+
+DeviceVector DeviceVector::getDevicesFromTypeAddr(
+ audio_devices_t type, String8 address) const
+{
+ DeviceVector devices;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mDeviceType == type) {
+ if (itemAt(i)->mAddress == address) {
+ devices.add(itemAt(i));
+ }
+ }
+ }
+ return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getDeviceFromName(const String8& name) const
+{
+ sp<DeviceDescriptor> device;
+ for (size_t i = 0; i < size(); i++) {
+ if (itemAt(i)->mName == name) {
+ device = itemAt(i);
+ break;
+ }
+ }
+ return device;
+}
+
+void DeviceDescriptor::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ dstConfig->config_mask = AUDIO_PORT_CONFIG_CHANNEL_MASK|AUDIO_PORT_CONFIG_GAIN;
+ if (srcConfig != NULL) {
+ dstConfig->config_mask |= srcConfig->config_mask;
+ }
+
+ AudioPortConfig::toAudioPortConfig(dstConfig, srcConfig);
+
+ dstConfig->id = mId;
+ dstConfig->role = audio_is_output_device(mDeviceType) ?
+ AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+ dstConfig->type = AUDIO_PORT_TYPE_DEVICE;
+ dstConfig->ext.device.type = mDeviceType;
+
+ //TODO Understand why this test is necessary. i.e. why at boot time does it crash
+ // without the test?
+ // This has been demonstrated to NOT be true (at start up)
+ // ALOG_ASSERT(mModule != NULL);
+ dstConfig->ext.device.hw_module = mModule != NULL ? mModule->mHandle : NULL;
+ strncpy(dstConfig->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+void DeviceDescriptor::toAudioPort(struct audio_port *port) const
+{
+ ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+ AudioPort::toAudioPort(port);
+ port->id = mId;
+ toAudioPortConfig(&port->active_config);
+ port->ext.device.type = mDeviceType;
+ port->ext.device.hw_module = mModule->mHandle;
+ strncpy(port->ext.device.address, mAddress.string(), AUDIO_DEVICE_MAX_ADDRESS_LEN);
+}
+
+status_t DeviceDescriptor::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sDevice %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ if (mId != 0) {
+ snprintf(buffer, SIZE, "%*s- id: %2d\n", spaces, "", mId);
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- type: %-48s\n", spaces, "",
+ ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mDeviceType));
+ result.append(buffer);
+ if (mAddress.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- address: %-32s\n", spaces, "", mAddress.string());
+ result.append(buffer);
+ }
+ write(fd, result.string(), result.size());
+ AudioPort::dump(fd, spaces);
+
+ return NO_ERROR;
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Devices.h b/services/audiopolicy/managerdefault/Devices.h
new file mode 100644
index 0000000..65e1416
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Devices.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class AudioPort;
+class AudioPortConfig;
+
+class DeviceDescriptor: public AudioPort, public AudioPortConfig
+{
+public:
+ DeviceDescriptor(const String8& name, audio_devices_t type);
+
+ virtual ~DeviceDescriptor() {}
+
+ bool equals(const sp<DeviceDescriptor>& other) const;
+
+ // AudioPortConfig
+ virtual sp<AudioPort> getAudioPort() const { return (AudioPort*) this; }
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const;
+
+ // AudioPort
+ virtual void loadGains(cnode *root);
+ 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_port_handle_t mId;
+
+ static String8 emptyNameStr;
+};
+
+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;
+ DeviceVector getDevicesFromTypeAddr(audio_devices_t type, String8 address)
+ const;
+
+private:
+ void refreshTypes();
+ audio_devices_t mDeviceTypes;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.cpp b/services/audiopolicy/managerdefault/Gains.cpp
new file mode 100644
index 0000000..4aca26d
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.cpp
@@ -0,0 +1,446 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Gains"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include "AudioPolicyManager.h"
+
+#include <math.h>
+
+namespace android {
+
+const VolumeCurvePoint
+ApmGains::sDefaultVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f}
+};
+
+
+const VolumeCurvePoint
+ApmGains::sDefaultMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sExtMediaSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -35.7f}, {33, -26.1f}, {66, -13.2f}, {100, 0.0f}
+};
+
+// AUDIO_STREAM_SYSTEM, AUDIO_STREAM_ENFORCED_AUDIBLE and AUDIO_STREAM_DTMF volume tracks
+// AUDIO_STREAM_RING on phones and AUDIO_STREAM_MUSIC on tablets.
+// AUDIO_STREAM_DTMF tracks AUDIO_STREAM_VOICE_CALL while in call (See AudioService.java).
+// The range is constrained between -24dB and -6dB over speaker and -30dB and -18dB over headset.
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT] = {
+ {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sHeadsetSystemVolumeCurve[ApmGains::VOLCNT] = {
+ {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sDefaultVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sLinearVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sSilentVolumeCurve[ApmGains::VOLCNT] = {
+ {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f}
+};
+
+const VolumeCurvePoint
+ApmGains::sFullScaleVolumeCurve[ApmGains::VOLCNT] = {
+ {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f}
+};
+
+const VolumeCurvePoint *ApmGains::sVolumeProfiles[AUDIO_STREAM_CNT]
+ [ApmGains::DEVICE_CATEGORY_CNT] = {
+ { // AUDIO_STREAM_VOICE_CALL
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_SYSTEM
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_RING
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_MUSIC
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ALARM
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_NOTIFICATION
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_BLUETOOTH_SCO
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ENFORCED_AUDIBLE
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_DTMF
+ ApmGains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_TTS
+ // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_ACCESSIBILITY
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_REROUTING
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+ { // AUDIO_STREAM_PATCH
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER
+ ApmGains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE
+ ApmGains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA
+ },
+};
+
+//static
+audio_devices_t ApmGains::getDeviceForVolume(audio_devices_t device)
+{
+ if (device == AUDIO_DEVICE_NONE) {
+ // this happens when forcing a route update and no track is active on an output.
+ // In this case the returned category is not important.
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (popcount(device) > 1) {
+ // Multiple device selection is either:
+ // - speaker + one other device: give priority to speaker in this case.
+ // - one A2DP device + another device: happens with duplicated output. In this case
+ // retain the device on the A2DP output as the other must not correspond to an active
+ // selection if not the speaker.
+ // - HDMI-CEC system audio mode only output: give priority to available item in order.
+ if (device & AUDIO_DEVICE_OUT_SPEAKER) {
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+ } else if (device & AUDIO_DEVICE_OUT_HDMI_ARC) {
+ device = AUDIO_DEVICE_OUT_HDMI_ARC;
+ } else if (device & AUDIO_DEVICE_OUT_AUX_LINE) {
+ device = AUDIO_DEVICE_OUT_AUX_LINE;
+ } else if (device & AUDIO_DEVICE_OUT_SPDIF) {
+ device = AUDIO_DEVICE_OUT_SPDIF;
+ } else {
+ device = (audio_devices_t)(device & AUDIO_DEVICE_OUT_ALL_A2DP);
+ }
+ }
+
+ /*SPEAKER_SAFE is an alias of SPEAKER for purposes of volume control*/
+ if (device == AUDIO_DEVICE_OUT_SPEAKER_SAFE)
+ device = AUDIO_DEVICE_OUT_SPEAKER;
+
+ ALOGW_IF(popcount(device) != 1,
+ "getDeviceForVolume() invalid device combination: %08x",
+ device);
+
+ return device;
+}
+
+//static
+ApmGains::device_category ApmGains::getDeviceCategory(audio_devices_t device)
+{
+ switch(getDeviceForVolume(device)) {
+ case AUDIO_DEVICE_OUT_EARPIECE:
+ return ApmGains::DEVICE_CATEGORY_EARPIECE;
+ case AUDIO_DEVICE_OUT_WIRED_HEADSET:
+ case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES:
+ return ApmGains::DEVICE_CATEGORY_HEADSET;
+ case AUDIO_DEVICE_OUT_LINE:
+ case AUDIO_DEVICE_OUT_AUX_DIGITAL:
+ /*USB? Remote submix?*/
+ return ApmGains::DEVICE_CATEGORY_EXT_MEDIA;
+ case AUDIO_DEVICE_OUT_SPEAKER:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
+ case AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER:
+ case AUDIO_DEVICE_OUT_USB_ACCESSORY:
+ case AUDIO_DEVICE_OUT_USB_DEVICE:
+ case AUDIO_DEVICE_OUT_REMOTE_SUBMIX:
+ default:
+ return ApmGains::DEVICE_CATEGORY_SPEAKER;
+ }
+}
+
+//static
+float ApmGains::volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi)
+{
+ ApmGains::device_category deviceCategory = ApmGains::getDeviceCategory(device);
+ const VolumeCurvePoint *curve = streamDesc.mVolumeCurve[deviceCategory];
+
+ // the volume index in the UI is relative to the min and max volume indices for this stream type
+ int nbSteps = 1 + curve[ApmGains::VOLMAX].mIndex -
+ curve[ApmGains::VOLMIN].mIndex;
+ int volIdx = (nbSteps * (indexInUi - streamDesc.mIndexMin)) /
+ (streamDesc.mIndexMax - streamDesc.mIndexMin);
+
+ // find what part of the curve this index volume belongs to, or if it's out of bounds
+ int segment = 0;
+ if (volIdx < curve[ApmGains::VOLMIN].mIndex) { // out of bounds
+ return 0.0f;
+ } else if (volIdx < curve[ApmGains::VOLKNEE1].mIndex) {
+ segment = 0;
+ } else if (volIdx < curve[ApmGains::VOLKNEE2].mIndex) {
+ segment = 1;
+ } else if (volIdx <= curve[ApmGains::VOLMAX].mIndex) {
+ segment = 2;
+ } else { // out of bounds
+ return 1.0f;
+ }
+
+ // linear interpolation in the attenuation table in dB
+ float decibels = curve[segment].mDBAttenuation +
+ ((float)(volIdx - curve[segment].mIndex)) *
+ ( (curve[segment+1].mDBAttenuation -
+ curve[segment].mDBAttenuation) /
+ ((float)(curve[segment+1].mIndex -
+ curve[segment].mIndex)) );
+
+ float amplification = exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 )
+
+ ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f] ampl=%.5f",
+ curve[segment].mIndex, volIdx,
+ curve[segment+1].mIndex,
+ curve[segment].mDBAttenuation,
+ decibels,
+ curve[segment+1].mDBAttenuation,
+ amplification);
+
+ return amplification;
+}
+
+
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+ mIndex = index;
+ mUseInChannelMask = useInChannelMask;
+ memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+ config->index = mIndex;
+ config->mode = mGain.mode;
+ config->channel_mask = mGain.channel_mask;
+ if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ config->values[0] = mGain.default_value;
+ } else {
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ config->values[i] = mGain.default_value;
+ }
+ }
+ if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ config->ramp_duration_ms = mGain.min_ramp_ms;
+ }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+ if ((config->mode & ~mGain.mode) != 0) {
+ return BAD_VALUE;
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+ if ((config->values[0] < mGain.min_value) ||
+ (config->values[0] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ } else {
+ if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+ return BAD_VALUE;
+ }
+ uint32_t numValues;
+ if (mUseInChannelMask) {
+ numValues = audio_channel_count_from_in_mask(config->channel_mask);
+ } else {
+ numValues = audio_channel_count_from_out_mask(config->channel_mask);
+ }
+ for (size_t i = 0; i < numValues; i++) {
+ if ((config->values[i] < mGain.min_value) ||
+ (config->values[i] > mGain.max_value)) {
+ return BAD_VALUE;
+ }
+ }
+ }
+ if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+ if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+ (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+ return BAD_VALUE;
+ }
+ }
+ return NO_ERROR;
+}
+
+void AudioGain::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sGain %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- mode: %08x\n", spaces, "", mGain.mode);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
+ result.append(buffer);
+
+ write(fd, result.string(), result.size());
+}
+
+
+// --- StreamDescriptor class implementation
+
+StreamDescriptor::StreamDescriptor()
+ : mIndexMin(0), mIndexMax(1), mCanBeMuted(true)
+{
+ mIndexCur.add(AUDIO_DEVICE_OUT_DEFAULT, 0);
+}
+
+int StreamDescriptor::getVolumeIndex(audio_devices_t device)
+{
+ device = ApmGains::getDeviceForVolume(device);
+ // there is always a valid entry for AUDIO_DEVICE_OUT_DEFAULT
+ if (mIndexCur.indexOfKey(device) < 0) {
+ device = AUDIO_DEVICE_OUT_DEFAULT;
+ }
+ return mIndexCur.valueFor(device);
+}
+
+void StreamDescriptor::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%s %02d %02d ",
+ mCanBeMuted ? "true " : "false", mIndexMin, mIndexMax);
+ result.append(buffer);
+ for (size_t i = 0; i < mIndexCur.size(); i++) {
+ snprintf(buffer, SIZE, "%04x : %02d, ",
+ mIndexCur.keyAt(i),
+ mIndexCur.valueAt(i));
+ result.append(buffer);
+ }
+ result.append("\n");
+
+ write(fd, result.string(), result.size());
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Gains.h b/services/audiopolicy/managerdefault/Gains.h
new file mode 100644
index 0000000..b4ab129
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Gains.h
@@ -0,0 +1,112 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class VolumeCurvePoint
+{
+public:
+ int mIndex;
+ float mDBAttenuation;
+};
+
+class StreamDescriptor;
+
+class ApmGains
+{
+public :
+ // 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};
+
+ // device categories used for volume curve management.
+ enum device_category {
+ DEVICE_CATEGORY_HEADSET,
+ DEVICE_CATEGORY_SPEAKER,
+ DEVICE_CATEGORY_EARPIECE,
+ DEVICE_CATEGORY_EXT_MEDIA,
+ DEVICE_CATEGORY_CNT
+ };
+
+ // returns the category the device belongs to with regard to volume curve management
+ static ApmGains::device_category getDeviceCategory(audio_devices_t device);
+
+ // extract one device relevant for volume control from multiple device selection
+ static audio_devices_t getDeviceForVolume(audio_devices_t device);
+
+ static float volIndexToAmpl(audio_devices_t device, const StreamDescriptor& streamDesc,
+ int indexInUi);
+
+ // default volume curve
+ static const VolumeCurvePoint sDefaultVolumeCurve[ApmGains::VOLCNT];
+ // default volume curve for media strategy
+ static const VolumeCurvePoint sDefaultMediaVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for non-media audio on ext media outputs (HDMI, Line, etc)
+ static const VolumeCurvePoint sExtMediaSystemVolumeCurve[ApmGains::VOLCNT];
+ // volume curve for media strategy on speakers
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[ApmGains::VOLCNT];
+ // volume curve for sonification strategy on speakers
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sHeadsetSystemVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sDefaultVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sLinearVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sSilentVolumeCurve[ApmGains::VOLCNT];
+ static const VolumeCurvePoint sFullScaleVolumeCurve[ApmGains::VOLCNT];
+ // default volume curves per stream and device category. See initializeVolumeCurves()
+ static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+
+class AudioGain: public RefBase
+{
+public:
+ AudioGain(int index, bool useInChannelMask);
+ virtual ~AudioGain() {}
+
+ void dump(int fd, int spaces, int index) const;
+
+ void getDefaultConfig(struct audio_gain_config *config);
+ status_t checkConfig(const struct audio_gain_config *config);
+ int mIndex;
+ struct audio_gain mGain;
+ bool mUseInChannelMask;
+};
+
+
+// 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[ApmGains::DEVICE_CATEGORY_CNT];
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.cpp b/services/audiopolicy/managerdefault/HwModule.cpp
new file mode 100644
index 0000000..a04bdc8
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.cpp
@@ -0,0 +1,279 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::HwModule"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+#include "audio_policy_conf.h"
+#include <hardware/audio.h>
+
+namespace android {
+
+HwModule::HwModule(const char *name)
+ : mName(strndup(name, AUDIO_HARDWARE_MODULE_ID_MAX_LEN)),
+ mHalVersion(AUDIO_DEVICE_API_VERSION_MIN), mHandle(0)
+{
+}
+
+HwModule::~HwModule()
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ mOutputProfiles[i]->mSupportedDevices.clear();
+ }
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ mInputProfiles[i]->mSupportedDevices.clear();
+ }
+ free((void *)mName);
+}
+
+status_t HwModule::loadInput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SINK, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadInChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseInputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadInput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadInput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadInput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadInput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadInput() adding input Supported Devices %04x",
+ profile->mSupportedDevices.types());
+
+ mInputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadOutput(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ sp<IOProfile> profile = new IOProfile(String8(root->name), AUDIO_PORT_ROLE_SOURCE, this);
+
+ while (node) {
+ if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
+ profile->loadSamplingRates((char *)node->value);
+ } else if (strcmp(node->name, FORMATS_TAG) == 0) {
+ profile->loadFormats((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ profile->loadOutChannels((char *)node->value);
+ } else if (strcmp(node->name, DEVICES_TAG) == 0) {
+ profile->mSupportedDevices.loadDevicesFromName((char *)node->value,
+ mDeclaredDevices);
+ } else if (strcmp(node->name, FLAGS_TAG) == 0) {
+ profile->mFlags = ConfigParsingUtils::parseOutputFlagNames((char *)node->value);
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ profile->loadGains(node);
+ }
+ node = node->next;
+ }
+ ALOGW_IF(profile->mSupportedDevices.isEmpty(),
+ "loadOutput() invalid supported devices");
+ ALOGW_IF(profile->mChannelMasks.size() == 0,
+ "loadOutput() invalid supported channel masks");
+ ALOGW_IF(profile->mSamplingRates.size() == 0,
+ "loadOutput() invalid supported sampling rates");
+ ALOGW_IF(profile->mFormats.size() == 0,
+ "loadOutput() invalid supported formats");
+ if (!profile->mSupportedDevices.isEmpty() &&
+ (profile->mChannelMasks.size() != 0) &&
+ (profile->mSamplingRates.size() != 0) &&
+ (profile->mFormats.size() != 0)) {
+
+ ALOGV("loadOutput() adding output Supported Devices %04x, mFlags %04x",
+ profile->mSupportedDevices.types(), profile->mFlags);
+
+ mOutputProfiles.add(profile);
+ return NO_ERROR;
+ } else {
+ return BAD_VALUE;
+ }
+}
+
+status_t HwModule::loadDevice(cnode *root)
+{
+ cnode *node = root->first_child;
+
+ audio_devices_t type = AUDIO_DEVICE_NONE;
+ while (node) {
+ if (strcmp(node->name, DEVICE_TYPE) == 0) {
+ type = ConfigParsingUtils::parseDeviceNames((char *)node->value);
+ break;
+ }
+ node = node->next;
+ }
+ if (type == AUDIO_DEVICE_NONE ||
+ (!audio_is_input_device(type) && !audio_is_output_device(type))) {
+ ALOGW("loadDevice() bad type %08x", type);
+ return BAD_VALUE;
+ }
+ sp<DeviceDescriptor> deviceDesc = new DeviceDescriptor(String8(root->name), type);
+ deviceDesc->mModule = this;
+
+ node = root->first_child;
+ while (node) {
+ if (strcmp(node->name, DEVICE_ADDRESS) == 0) {
+ deviceDesc->mAddress = String8((char *)node->value);
+ } else if (strcmp(node->name, CHANNELS_TAG) == 0) {
+ if (audio_is_input_device(type)) {
+ deviceDesc->loadInChannels((char *)node->value);
+ } else {
+ deviceDesc->loadOutChannels((char *)node->value);
+ }
+ } else if (strcmp(node->name, GAINS_TAG) == 0) {
+ deviceDesc->loadGains(node);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadDevice() adding device name %s type %08x address %s",
+ deviceDesc->mName.string(), type, deviceDesc->mAddress.string());
+
+ mDeclaredDevices.add(deviceDesc);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SOURCE, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ mOutputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeOutputProfile(String8 name)
+{
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ if (mOutputProfiles[i]->mName == name) {
+ mOutputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+status_t HwModule::addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address)
+{
+ sp<IOProfile> profile = new IOProfile(name, AUDIO_PORT_ROLE_SINK, this);
+
+ profile->mSamplingRates.add(config->sample_rate);
+ profile->mChannelMasks.add(config->channel_mask);
+ profile->mFormats.add(config->format);
+
+ sp<DeviceDescriptor> devDesc = new DeviceDescriptor(name, device);
+ devDesc->mAddress = address;
+ profile->mSupportedDevices.add(devDesc);
+
+ ALOGV("addInputProfile() name %s rate %d mask 0x08", name.string(), config->sample_rate, config->channel_mask);
+
+ mInputProfiles.add(profile);
+
+ return NO_ERROR;
+}
+
+status_t HwModule::removeInputProfile(String8 name)
+{
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ if (mInputProfiles[i]->mName == name) {
+ mInputProfiles.removeAt(i);
+ break;
+ }
+ }
+
+ return NO_ERROR;
+}
+
+
+void HwModule::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, " - name: %s\n", mName);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - handle: %d\n", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - version: %u.%u\n", mHalVersion >> 8, mHalVersion & 0xFF);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ if (mOutputProfiles.size()) {
+ write(fd, " - outputs:\n", strlen(" - outputs:\n"));
+ for (size_t i = 0; i < mOutputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " output %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mOutputProfiles[i]->dump(fd);
+ }
+ }
+ if (mInputProfiles.size()) {
+ write(fd, " - inputs:\n", strlen(" - inputs:\n"));
+ for (size_t i = 0; i < mInputProfiles.size(); i++) {
+ snprintf(buffer, SIZE, " input %zu:\n", i);
+ write(fd, buffer, strlen(buffer));
+ mInputProfiles[i]->dump(fd);
+ }
+ }
+ if (mDeclaredDevices.size()) {
+ write(fd, " - devices:\n", strlen(" - devices:\n"));
+ for (size_t i = 0; i < mDeclaredDevices.size(); i++) {
+ mDeclaredDevices[i]->dump(fd, 4, i);
+ }
+ }
+}
+
+} //namespace android
diff --git a/services/audiopolicy/managerdefault/HwModule.h b/services/audiopolicy/managerdefault/HwModule.h
new file mode 100644
index 0000000..f814dd9
--- /dev/null
+++ b/services/audiopolicy/managerdefault/HwModule.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule : public RefBase
+{
+public:
+ HwModule(const char *name);
+ ~HwModule();
+
+ status_t loadOutput(cnode *root);
+ status_t loadInput(cnode *root);
+ status_t loadDevice(cnode *root);
+
+ status_t addOutputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeOutputProfile(String8 name);
+ status_t addInputProfile(String8 name, const audio_config_t *config,
+ audio_devices_t device, String8 address);
+ status_t removeInputProfile(String8 name);
+
+ void dump(int fd);
+
+ const char *const mName; // base name of the audio HW module (primary, a2dp ...)
+ uint32_t mHalVersion; // audio HAL API version
+ 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
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.cpp b/services/audiopolicy/managerdefault/IOProfile.cpp
new file mode 100644
index 0000000..538ac1a
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.cpp
@@ -0,0 +1,139 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::IOProfile"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+namespace android {
+
+IOProfile::IOProfile(const String8& name, audio_port_role_t role,
+ const sp<HwModule>& module)
+ : AudioPort(name, AUDIO_PORT_TYPE_MIX, role, module)
+{
+}
+
+IOProfile::~IOProfile()
+{
+}
+
+// checks if the IO profile is compatible with specified parameters.
+// Sampling rate, format and channel mask must be specified in order to
+// get a valid a match
+bool IOProfile::isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const
+{
+ const bool isPlaybackThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SOURCE;
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ ALOG_ASSERT(isPlaybackThread != isRecordThread);
+
+ if (device != AUDIO_DEVICE_NONE && mSupportedDevices.getDevice(device, address) == 0) {
+ return false;
+ }
+
+ if (samplingRate == 0) {
+ return false;
+ }
+ uint32_t myUpdatedSamplingRate = samplingRate;
+ if (isPlaybackThread && checkExactSamplingRate(samplingRate) != NO_ERROR) {
+ return false;
+ }
+ if (isRecordThread && checkCompatibleSamplingRate(samplingRate, &myUpdatedSamplingRate) !=
+ NO_ERROR) {
+ return false;
+ }
+
+ if (!audio_is_valid_format(format) || checkFormat(format) != NO_ERROR) {
+ return false;
+ }
+
+ if (isPlaybackThread && (!audio_is_output_channel(channelMask) ||
+ checkExactChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+ if (isRecordThread && (!audio_is_input_channel(channelMask) ||
+ checkCompatibleChannelMask(channelMask) != NO_ERROR)) {
+ return false;
+ }
+
+ if (isPlaybackThread && (mFlags & flags) != flags) {
+ return false;
+ }
+ // The only input flag that is allowed to be different is the fast flag.
+ // An existing fast stream is compatible with a normal track request.
+ // An existing normal stream is compatible with a fast track request,
+ // but the fast request will be denied by AudioFlinger and converted to normal track.
+ if (isRecordThread && ((mFlags ^ flags) &
+ ~AUDIO_INPUT_FLAG_FAST)) {
+ return false;
+ }
+
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = myUpdatedSamplingRate;
+ }
+ return true;
+}
+
+void IOProfile::dump(int fd)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ AudioPort::dump(fd, 4);
+
+ snprintf(buffer, SIZE, " - flags: 0x%04x\n", mFlags);
+ result.append(buffer);
+ snprintf(buffer, SIZE, " - devices:\n");
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ for (size_t i = 0; i < mSupportedDevices.size(); i++) {
+ mSupportedDevices[i]->dump(fd, 6, i);
+ }
+}
+
+void IOProfile::log()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ ALOGV(" - sampling rates: ");
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ ALOGV(" %d", mSamplingRates[i]);
+ }
+
+ ALOGV(" - channel masks: ");
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV(" 0x%04x", mChannelMasks[i]);
+ }
+
+ ALOGV(" - formats: ");
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ ALOGV(" 0x%08x", mFormats[i]);
+ }
+
+ ALOGV(" - devices: 0x%04x\n", mSupportedDevices.types());
+ ALOGV(" - flags: 0x%04x\n", mFlags);
+}
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/IOProfile.h b/services/audiopolicy/managerdefault/IOProfile.h
new file mode 100644
index 0000000..3317969
--- /dev/null
+++ b/services/audiopolicy/managerdefault/IOProfile.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule;
+
+// 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, const sp<HwModule>& module);
+ virtual ~IOProfile();
+
+ // This method is used for both output and input.
+ // If parameter updatedSamplingRate is non-NULL, it is assigned the actual sample rate.
+ // For input, flags is interpreted as audio_input_flags_t.
+ // TODO: merge audio_output_flags_t and audio_input_flags_t.
+ bool isCompatibleProfile(audio_devices_t device,
+ String8 address,
+ uint32_t samplingRate,
+ uint32_t *updatedSamplingRate,
+ audio_format_t format,
+ audio_channel_mask_t channelMask,
+ uint32_t flags) const;
+
+ void dump(int fd);
+ void log();
+
+ DeviceVector mSupportedDevices; // supported devices
+ // (devices this output can be routed to)
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.cpp b/services/audiopolicy/managerdefault/Ports.cpp
new file mode 100644
index 0000000..3e55cee
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.cpp
@@ -0,0 +1,844 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM::Ports"
+//#define LOG_NDEBUG 0
+
+#include "AudioPolicyManager.h"
+
+#include "audio_policy_conf.h"
+
+namespace android {
+
+// --- AudioPort class implementation
+
+AudioPort::AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module) :
+ mName(name), mType(type), mRole(role), mModule(module), mFlags(0), mId(0)
+{
+ mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) ||
+ ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK));
+}
+
+void AudioPort::attach(const sp<HwModule>& module) {
+ mId = AudioPolicyManager::nextUniqueId();
+ mModule = module;
+}
+
+void AudioPort::toAudioPort(struct audio_port *port) const
+{
+ port->role = mRole;
+ port->type = mType;
+ strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
+ unsigned int i;
+ for (i = 0; i < mSamplingRates.size() && i < AUDIO_PORT_MAX_SAMPLING_RATES; i++) {
+ if (mSamplingRates[i] != 0) {
+ port->sample_rates[i] = mSamplingRates[i];
+ }
+ }
+ port->num_sample_rates = i;
+ for (i = 0; i < mChannelMasks.size() && i < AUDIO_PORT_MAX_CHANNEL_MASKS; i++) {
+ if (mChannelMasks[i] != 0) {
+ port->channel_masks[i] = mChannelMasks[i];
+ }
+ }
+ port->num_channel_masks = i;
+ for (i = 0; i < mFormats.size() && i < AUDIO_PORT_MAX_FORMATS; i++) {
+ if (mFormats[i] != 0) {
+ port->formats[i] = mFormats[i];
+ }
+ }
+ port->num_formats = i;
+
+ ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+ for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
+ port->gains[i] = mGains[i]->mGain;
+ }
+ port->num_gains = i;
+}
+
+void AudioPort::importAudioPort(const sp<AudioPort> port) {
+ for (size_t k = 0 ; k < port->mSamplingRates.size() ; k++) {
+ const uint32_t rate = port->mSamplingRates.itemAt(k);
+ if (rate != 0) { // skip "dynamic" rates
+ bool hasRate = false;
+ for (size_t l = 0 ; l < mSamplingRates.size() ; l++) {
+ if (rate == mSamplingRates.itemAt(l)) {
+ hasRate = true;
+ break;
+ }
+ }
+ if (!hasRate) { // never import a sampling rate twice
+ mSamplingRates.add(rate);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mChannelMasks.size() ; k++) {
+ const audio_channel_mask_t mask = port->mChannelMasks.itemAt(k);
+ if (mask != 0) { // skip "dynamic" masks
+ bool hasMask = false;
+ for (size_t l = 0 ; l < mChannelMasks.size() ; l++) {
+ if (mask == mChannelMasks.itemAt(l)) {
+ hasMask = true;
+ break;
+ }
+ }
+ if (!hasMask) { // never import a channel mask twice
+ mChannelMasks.add(mask);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mFormats.size() ; k++) {
+ const audio_format_t format = port->mFormats.itemAt(k);
+ if (format != 0) { // skip "dynamic" formats
+ bool hasFormat = false;
+ for (size_t l = 0 ; l < mFormats.size() ; l++) {
+ if (format == mFormats.itemAt(l)) {
+ hasFormat = true;
+ break;
+ }
+ }
+ if (!hasFormat) { // never import a channel mask twice
+ mFormats.add(format);
+ }
+ }
+ }
+ for (size_t k = 0 ; k < port->mGains.size() ; k++) {
+ sp<AudioGain> gain = port->mGains.itemAt(k);
+ if (gain != 0) {
+ bool hasGain = false;
+ for (size_t l = 0 ; l < mGains.size() ; l++) {
+ if (gain == mGains.itemAt(l)) {
+ hasGain = true;
+ break;
+ }
+ }
+ if (!hasGain) { // never import a gain twice
+ mGains.add(gain);
+ }
+ }
+ }
+}
+
+void AudioPort::clearCapabilities() {
+ mChannelMasks.clear();
+ mFormats.clear();
+ mSamplingRates.clear();
+ mGains.clear();
+}
+
+void AudioPort::loadSamplingRates(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mSamplingRates indicates the supported sampling
+ // rates should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mSamplingRates.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ uint32_t rate = atoi(str);
+ if (rate != 0) {
+ ALOGV("loadSamplingRates() adding rate %d", rate);
+ mSamplingRates.add(rate);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadFormats(char *name)
+{
+ char *str = strtok(name, "|");
+
+ // by convention, "0' in the first entry in mFormats indicates the supported formats
+ // should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mFormats.add(AUDIO_FORMAT_DEFAULT);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_format_t format = (audio_format_t)ConfigParsingUtils::stringToEnum(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ str);
+ if (format != AUDIO_FORMAT_DEFAULT) {
+ mFormats.add(format);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadInChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadInChannels() %s", name);
+
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ ALOGV("loadInChannels() adding channelMask %04x", channelMask);
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+}
+
+void AudioPort::loadOutChannels(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadOutChannels() %s", name);
+
+ // by convention, "0' in the first entry in mChannelMasks indicates the supported channel
+ // masks should be read from the output stream after it is opened for the first time
+ if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0) {
+ mChannelMasks.add(0);
+ return;
+ }
+
+ while (str != NULL) {
+ audio_channel_mask_t channelMask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ str);
+ if (channelMask != 0) {
+ mChannelMasks.add(channelMask);
+ }
+ str = strtok(NULL, "|");
+ }
+ return;
+}
+
+audio_gain_mode_t AudioPort::loadGainMode(char *name)
+{
+ const char *str = strtok(name, "|");
+
+ ALOGV("loadGainMode() %s", name);
+ audio_gain_mode_t mode = 0;
+ while (str != NULL) {
+ mode |= (audio_gain_mode_t)ConfigParsingUtils::stringToEnum(sGainModeNameToEnumTable,
+ ARRAY_SIZE(sGainModeNameToEnumTable),
+ str);
+ str = strtok(NULL, "|");
+ }
+ return mode;
+}
+
+void AudioPort::loadGain(cnode *root, int index)
+{
+ cnode *node = root->first_child;
+
+ sp<AudioGain> gain = new AudioGain(index, mUseInChannelMask);
+
+ while (node) {
+ if (strcmp(node->name, GAIN_MODE) == 0) {
+ gain->mGain.mode = loadGainMode((char *)node->value);
+ } else if (strcmp(node->name, GAIN_CHANNELS) == 0) {
+ if (mUseInChannelMask) {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sInChannelsNameToEnumTable,
+ ARRAY_SIZE(sInChannelsNameToEnumTable),
+ (char *)node->value);
+ } else {
+ gain->mGain.channel_mask =
+ (audio_channel_mask_t)ConfigParsingUtils::stringToEnum(sOutChannelsNameToEnumTable,
+ ARRAY_SIZE(sOutChannelsNameToEnumTable),
+ (char *)node->value);
+ }
+ } else if (strcmp(node->name, GAIN_MIN_VALUE) == 0) {
+ gain->mGain.min_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_VALUE) == 0) {
+ gain->mGain.max_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_DEFAULT_VALUE) == 0) {
+ gain->mGain.default_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_STEP_VALUE) == 0) {
+ gain->mGain.step_value = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MIN_RAMP_MS) == 0) {
+ gain->mGain.min_ramp_ms = atoi((char *)node->value);
+ } else if (strcmp(node->name, GAIN_MAX_RAMP_MS) == 0) {
+ gain->mGain.max_ramp_ms = atoi((char *)node->value);
+ }
+ node = node->next;
+ }
+
+ ALOGV("loadGain() adding new gain mode %08x channel mask %08x min mB %d max mB %d",
+ gain->mGain.mode, gain->mGain.channel_mask, gain->mGain.min_value, gain->mGain.max_value);
+
+ if (gain->mGain.mode == 0) {
+ return;
+ }
+ mGains.add(gain);
+}
+
+void AudioPort::loadGains(cnode *root)
+{
+ cnode *node = root->first_child;
+ int index = 0;
+ while (node) {
+ ALOGV("loadGains() loading gain %s", node->name);
+ loadGain(node, index++);
+ node = node->next;
+ }
+}
+
+status_t AudioPort::checkExactSamplingRate(uint32_t samplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if (mSamplingRates[i] == samplingRate) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const
+{
+ if (mSamplingRates.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ // Search for the closest supported sampling rate that is above (preferred)
+ // or below (acceptable) the desired sampling rate, within a permitted ratio.
+ // The sampling rates do not need to be sorted in ascending order.
+ ssize_t maxBelow = -1;
+ ssize_t minAbove = -1;
+ uint32_t candidate;
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ candidate = mSamplingRates[i];
+ if (candidate == samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ // candidate < desired
+ if (candidate < samplingRate) {
+ if (maxBelow < 0 || candidate > mSamplingRates[maxBelow]) {
+ maxBelow = i;
+ }
+ // candidate > desired
+ } else {
+ if (minAbove < 0 || candidate < mSamplingRates[minAbove]) {
+ minAbove = i;
+ }
+ }
+ }
+ // This uses hard-coded knowledge about AudioFlinger resampling ratios.
+ // TODO Move these assumptions out.
+ static const uint32_t kMaxDownSampleRatio = 6; // beyond this aliasing occurs
+ static const uint32_t kMaxUpSampleRatio = 256; // beyond this sample rate inaccuracies occur
+ // due to approximation by an int32_t of the
+ // phase increments
+ // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
+ if (minAbove >= 0) {
+ candidate = mSamplingRates[minAbove];
+ if (candidate / kMaxDownSampleRatio <= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // But if we have to up-sample from a lower sampling rate, that's OK.
+ if (maxBelow >= 0) {
+ candidate = mSamplingRates[maxBelow];
+ if (candidate * kMaxUpSampleRatio >= samplingRate) {
+ if (updatedSamplingRate != NULL) {
+ *updatedSamplingRate = candidate;
+ }
+ return NO_ERROR;
+ }
+ }
+ // leave updatedSamplingRate unmodified
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkExactChannelMask(audio_channel_mask_t channelMask) const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ if (mChannelMasks[i] == channelMask) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkCompatibleChannelMask(audio_channel_mask_t channelMask)
+ const
+{
+ if (mChannelMasks.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ const bool isRecordThread = mType == AUDIO_PORT_TYPE_MIX && mRole == AUDIO_PORT_ROLE_SINK;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ // FIXME Does not handle multi-channel automatic conversions yet
+ audio_channel_mask_t supported = mChannelMasks[i];
+ if (supported == channelMask) {
+ return NO_ERROR;
+ }
+ if (isRecordThread) {
+ // This uses hard-coded knowledge that AudioFlinger can silently down-mix and up-mix.
+ // FIXME Abstract this out to a table.
+ if (((supported == AUDIO_CHANNEL_IN_FRONT_BACK || supported == AUDIO_CHANNEL_IN_STEREO)
+ && channelMask == AUDIO_CHANNEL_IN_MONO) ||
+ (supported == AUDIO_CHANNEL_IN_MONO && (channelMask == AUDIO_CHANNEL_IN_FRONT_BACK
+ || channelMask == AUDIO_CHANNEL_IN_STEREO))) {
+ return NO_ERROR;
+ }
+ }
+ }
+ return BAD_VALUE;
+}
+
+status_t AudioPort::checkFormat(audio_format_t format) const
+{
+ if (mFormats.isEmpty()) {
+ return NO_ERROR;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if (mFormats[i] == format) {
+ return NO_ERROR;
+ }
+ }
+ return BAD_VALUE;
+}
+
+
+uint32_t AudioPort::pickSamplingRate() const
+{
+ // special case for uninitialized dynamic profile
+ if (mSamplingRates.size() == 1 && mSamplingRates[0] == 0) {
+ return 0;
+ }
+
+ // For direct outputs, pick minimum sampling rate: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t samplingRate = UINT_MAX;
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] < samplingRate) && (mSamplingRates[i] > 0)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return (samplingRate == UINT_MAX) ? 0 : samplingRate;
+ }
+
+ uint32_t samplingRate = 0;
+ uint32_t maxRate = MAX_MIXER_SAMPLING_RATE;
+
+ // For mixed output and inputs, use max mixer sampling rates. Do not
+ // limit sampling rate otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxRate = UINT_MAX;
+ }
+ for (size_t i = 0; i < mSamplingRates.size(); i ++) {
+ if ((mSamplingRates[i] > samplingRate) && (mSamplingRates[i] <= maxRate)) {
+ samplingRate = mSamplingRates[i];
+ }
+ }
+ return samplingRate;
+}
+
+audio_channel_mask_t AudioPort::pickChannelMask() const
+{
+ // special case for uninitialized dynamic profile
+ if (mChannelMasks.size() == 1 && mChannelMasks[0] == 0) {
+ return AUDIO_CHANNEL_NONE;
+ }
+ audio_channel_mask_t channelMask = AUDIO_CHANNEL_NONE;
+
+ // For direct outputs, pick minimum channel count: this helps ensuring that the
+ // channel count / sampling rate combination chosen will be supported by the connected
+ // sink
+ if ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))) {
+ uint32_t channelCount = UINT_MAX;
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount < channelCount) && (cnlCount > 0)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+ }
+
+ uint32_t channelCount = 0;
+ uint32_t maxCount = MAX_MIXER_CHANNEL_COUNT;
+
+ // For mixed output and inputs, use max mixer channel count. Do not
+ // limit channel count otherwise
+ if (mType != AUDIO_PORT_TYPE_MIX) {
+ maxCount = UINT_MAX;
+ }
+ for (size_t i = 0; i < mChannelMasks.size(); i ++) {
+ uint32_t cnlCount;
+ if (mUseInChannelMask) {
+ cnlCount = audio_channel_count_from_in_mask(mChannelMasks[i]);
+ } else {
+ cnlCount = audio_channel_count_from_out_mask(mChannelMasks[i]);
+ }
+ if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
+ channelMask = mChannelMasks[i];
+ channelCount = cnlCount;
+ }
+ }
+ return channelMask;
+}
+
+/* format in order of increasing preference */
+const audio_format_t AudioPort::sPcmFormatCompareTable[] = {
+ AUDIO_FORMAT_DEFAULT,
+ AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_FORMAT_PCM_8_24_BIT,
+ AUDIO_FORMAT_PCM_24_BIT_PACKED,
+ AUDIO_FORMAT_PCM_32_BIT,
+ AUDIO_FORMAT_PCM_FLOAT,
+};
+
+int AudioPort::compareFormats(audio_format_t format1,
+ audio_format_t format2)
+{
+ // NOTE: AUDIO_FORMAT_INVALID is also considered not PCM and will be compared equal to any
+ // compressed format and better than any PCM format. This is by design of pickFormat()
+ if (!audio_is_linear_pcm(format1)) {
+ if (!audio_is_linear_pcm(format2)) {
+ return 0;
+ }
+ return 1;
+ }
+ if (!audio_is_linear_pcm(format2)) {
+ return -1;
+ }
+
+ int index1 = -1, index2 = -1;
+ for (size_t i = 0;
+ (i < ARRAY_SIZE(sPcmFormatCompareTable)) && ((index1 == -1) || (index2 == -1));
+ i ++) {
+ if (sPcmFormatCompareTable[i] == format1) {
+ index1 = i;
+ }
+ if (sPcmFormatCompareTable[i] == format2) {
+ index2 = i;
+ }
+ }
+ // format1 not found => index1 < 0 => format2 > format1
+ // format2 not found => index2 < 0 => format2 < format1
+ return index1 - index2;
+}
+
+audio_format_t AudioPort::pickFormat() const
+{
+ // special case for uninitialized dynamic profile
+ if (mFormats.size() == 1 && mFormats[0] == 0) {
+ return AUDIO_FORMAT_DEFAULT;
+ }
+
+ audio_format_t format = AUDIO_FORMAT_DEFAULT;
+ audio_format_t bestFormat =
+ AudioPort::sPcmFormatCompareTable[
+ ARRAY_SIZE(AudioPort::sPcmFormatCompareTable) - 1];
+ // For mixed output and inputs, use best mixer output format. Do not
+ // limit format otherwise
+ if ((mType != AUDIO_PORT_TYPE_MIX) ||
+ ((mRole == AUDIO_PORT_ROLE_SOURCE) &&
+ (((mFlags & (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) != 0)))) {
+ bestFormat = AUDIO_FORMAT_INVALID;
+ }
+
+ for (size_t i = 0; i < mFormats.size(); i ++) {
+ if ((compareFormats(mFormats[i], format) > 0) &&
+ (compareFormats(mFormats[i], bestFormat) <= 0)) {
+ format = mFormats[i];
+ }
+ }
+ return format;
+}
+
+status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig,
+ int index) const
+{
+ if (index < 0 || (size_t)index >= mGains.size()) {
+ return BAD_VALUE;
+ }
+ return mGains[index]->checkConfig(gainConfig);
+}
+
+void AudioPort::dump(int fd, int spaces) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ if (mName.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- name: %s\n", spaces, "", mName.string());
+ result.append(buffer);
+ }
+
+ if (mSamplingRates.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- sampling rates: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mSamplingRates.size(); i++) {
+ if (i == 0 && mSamplingRates[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%d", mSamplingRates[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mSamplingRates.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mChannelMasks.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- channel masks: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mChannelMasks.size(); i++) {
+ ALOGV("AudioPort::dump mChannelMasks %zu %08x", i, mChannelMasks[i]);
+
+ if (i == 0 && mChannelMasks[i] == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "0x%04x", mChannelMasks[i]);
+ }
+ result.append(buffer);
+ result.append(i == (mChannelMasks.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+
+ if (mFormats.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- formats: ", spaces, "");
+ result.append(buffer);
+ for (size_t i = 0; i < mFormats.size(); i++) {
+ const char *formatStr = ConfigParsingUtils::enumToString(sFormatNameToEnumTable,
+ ARRAY_SIZE(sFormatNameToEnumTable),
+ mFormats[i]);
+ if (i == 0 && strcmp(formatStr, "") == 0) {
+ snprintf(buffer, SIZE, "Dynamic");
+ } else {
+ snprintf(buffer, SIZE, "%s", formatStr);
+ }
+ result.append(buffer);
+ result.append(i == (mFormats.size() - 1) ? "" : ", ");
+ }
+ result.append("\n");
+ }
+ write(fd, result.string(), result.size());
+ if (mGains.size() != 0) {
+ snprintf(buffer, SIZE, "%*s- gains:\n", spaces, "");
+ write(fd, buffer, strlen(buffer) + 1);
+ result.append(buffer);
+ for (size_t i = 0; i < mGains.size(); i++) {
+ mGains[i]->dump(fd, spaces + 2, i);
+ }
+ }
+}
+
+
+// --- AudioPortConfig class implementation
+
+AudioPortConfig::AudioPortConfig()
+{
+ mSamplingRate = 0;
+ mChannelMask = AUDIO_CHANNEL_NONE;
+ mFormat = AUDIO_FORMAT_INVALID;
+ mGain.index = -1;
+}
+
+status_t AudioPortConfig::applyAudioPortConfig(
+ const struct audio_port_config *config,
+ struct audio_port_config *backupConfig)
+{
+ struct audio_port_config localBackupConfig;
+ status_t status = NO_ERROR;
+
+ localBackupConfig.config_mask = config->config_mask;
+ toAudioPortConfig(&localBackupConfig);
+
+ sp<AudioPort> audioport = getAudioPort();
+ if (audioport == 0) {
+ status = NO_INIT;
+ goto exit;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ status = audioport->checkExactSamplingRate(config->sample_rate);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mSamplingRate = config->sample_rate;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ status = audioport->checkExactChannelMask(config->channel_mask);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mChannelMask = config->channel_mask;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ status = audioport->checkFormat(config->format);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mFormat = config->format;
+ }
+ if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ status = audioport->checkGain(&config->gain, config->gain.index);
+ if (status != NO_ERROR) {
+ goto exit;
+ }
+ mGain = config->gain;
+ }
+
+exit:
+ if (status != NO_ERROR) {
+ applyAudioPortConfig(&localBackupConfig);
+ }
+ if (backupConfig != NULL) {
+ *backupConfig = localBackupConfig;
+ }
+ return status;
+}
+
+void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig) const
+{
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+ dstConfig->sample_rate = mSamplingRate;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
+ dstConfig->sample_rate = srcConfig->sample_rate;
+ }
+ } else {
+ dstConfig->sample_rate = 0;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+ dstConfig->channel_mask = mChannelMask;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
+ dstConfig->channel_mask = srcConfig->channel_mask;
+ }
+ } else {
+ dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+ dstConfig->format = mFormat;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
+ dstConfig->format = srcConfig->format;
+ }
+ } else {
+ dstConfig->format = AUDIO_FORMAT_INVALID;
+ }
+ if (dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) {
+ dstConfig->gain = mGain;
+ if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_GAIN)) {
+ dstConfig->gain = srcConfig->gain;
+ }
+ } else {
+ dstConfig->gain.index = -1;
+ }
+ if (dstConfig->gain.index != -1) {
+ dstConfig->config_mask |= AUDIO_PORT_CONFIG_GAIN;
+ } else {
+ dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
+ }
+}
+
+
+// --- AudioPatch class implementation
+
+AudioPatch::AudioPatch(audio_patch_handle_t handle,
+ const struct audio_patch *patch, uid_t uid) :
+ mHandle(handle), mPatch(*patch), mUid(uid), mAfPatchHandle(0)
+{}
+
+status_t AudioPatch::dump(int fd, int spaces, int index) const
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sources; i++) {
+ if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sources[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sources[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+ snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
+ result.append(buffer);
+ for (size_t i = 0; i < mPatch.num_sinks; i++) {
+ if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
+ snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
+ mPatch.sinks[i].id, ConfigParsingUtils::enumToString(sDeviceNameToEnumTable,
+ ARRAY_SIZE(sDeviceNameToEnumTable),
+ mPatch.sinks[i].ext.device.type));
+ } else {
+ snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
+ mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
+ }
+ result.append(buffer);
+ }
+
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+
+}; // namespace android
diff --git a/services/audiopolicy/managerdefault/Ports.h b/services/audiopolicy/managerdefault/Ports.h
new file mode 100644
index 0000000..f6e0e93
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Ports.h
@@ -0,0 +1,122 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+namespace android {
+
+class HwModule;
+
+class AudioPort: public virtual RefBase
+{
+public:
+ AudioPort(const String8& name, audio_port_type_t type,
+ audio_port_role_t role, const sp<HwModule>& module);
+ virtual ~AudioPort() {}
+
+ audio_port_handle_t getHandle() { return mId; }
+
+ void attach(const sp<HwModule>& module);
+ bool isAttached() { return mId != 0; }
+
+ virtual void toAudioPort(struct audio_port *port) const;
+
+ void importAudioPort(const sp<AudioPort> port);
+ void clearCapabilities();
+
+ 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, int index);
+ virtual void loadGains(cnode *root);
+
+ // searches for an exact match
+ status_t checkExactSamplingRate(uint32_t samplingRate) const;
+ // searches for a compatible match, and returns the best match via updatedSamplingRate
+ status_t checkCompatibleSamplingRate(uint32_t samplingRate,
+ uint32_t *updatedSamplingRate) const;
+ // searches for an exact match
+ status_t checkExactChannelMask(audio_channel_mask_t channelMask) const;
+ // searches for a compatible match, currently implemented for input channel masks only
+ status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask) const;
+ status_t checkFormat(audio_format_t format) const;
+ status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
+
+ uint32_t pickSamplingRate() const;
+ audio_channel_mask_t pickChannelMask() const;
+ audio_format_t pickFormat() const;
+
+ static const audio_format_t sPcmFormatCompareTable[];
+ static int compareFormats(audio_format_t format1, audio_format_t format2);
+
+ void dump(int fd, int spaces) const;
+
+ String8 mName;
+ audio_port_type_t mType;
+ audio_port_role_t mRole;
+ bool mUseInChannelMask;
+ // 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
+ sp<HwModule> mModule; // audio HW module exposing this I/O stream
+ uint32_t mFlags; // attribute flags (e.g primary output,
+ // direct output...).
+
+
+protected:
+ //TODO - clarify the role of mId in this case, both an "attached" indicator
+ // and a unique ID for identifying a port to the (upcoming) selection API,
+ // and its relationship to the mId in AudioOutputDescriptor and AudioInputDescriptor.
+ audio_port_handle_t mId;
+};
+
+class AudioPortConfig: public virtual RefBase
+{
+public:
+ AudioPortConfig();
+ virtual ~AudioPortConfig() {}
+
+ status_t applyAudioPortConfig(const struct audio_port_config *config,
+ struct audio_port_config *backupConfig = NULL);
+ virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+ const struct audio_port_config *srcConfig = NULL) const = 0;
+ virtual sp<AudioPort> getAudioPort() const = 0;
+ uint32_t mSamplingRate;
+ audio_format_t mFormat;
+ audio_channel_mask_t mChannelMask;
+ struct audio_gain_config mGain;
+};
+
+
+class AudioPatch: public RefBase
+{
+public:
+ AudioPatch(audio_patch_handle_t handle, const struct audio_patch *patch, uid_t uid);
+
+ status_t dump(int fd, int spaces, int index) const;
+
+ audio_patch_handle_t mHandle;
+ struct audio_patch mPatch;
+ uid_t mUid;
+ audio_patch_handle_t mAfPatchHandle;
+};
+
+}; // namespace android
diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/managerdefault/audio_policy_conf.h
similarity index 100%
rename from services/audiopolicy/audio_policy_conf.h
rename to services/audiopolicy/managerdefault/audio_policy_conf.h
diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImpl.cpp
rename to services/audiopolicy/service/AudioPolicyClientImpl.cpp
diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyClientImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.cpp
rename to services/audiopolicy/service/AudioPolicyEffects.cpp
diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
similarity index 100%
rename from services/audiopolicy/AudioPolicyEffects.h
rename to services/audiopolicy/service/AudioPolicyEffects.h
diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyInterfaceImpl.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp
rename to services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp
diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
similarity index 100%
rename from services/audiopolicy/AudioPolicyService.cpp
rename to services/audiopolicy/service/AudioPolicyService.cpp
diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
similarity index 99%
rename from services/audiopolicy/AudioPolicyService.h
rename to services/audiopolicy/service/AudioPolicyService.h
index 7c2b59d..0378384 100644
--- a/services/audiopolicy/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -35,7 +35,7 @@
#include <hardware_legacy/AudioPolicyInterface.h>
#endif
#include "AudioPolicyEffects.h"
-#include "AudioPolicyManager.h"
+#include "managerdefault/AudioPolicyManager.h"
namespace android {