diff options
Diffstat (limited to 'services/audiopolicy')
55 files changed, 8001 insertions, 5037 deletions
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk index 188fc89..d4ce86a 100644 --- a/services/audiopolicy/Android.mk +++ b/services/audiopolicy/Android.mk @@ -3,25 +3,27 @@ LOCAL_PATH:= $(call my-dir) 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 := \ $(TOPDIR)frameworks/av/services/audioflinger \ $(call include-path-for, audio-effects) \ - $(call include-path-for, audio-utils) + $(call include-path-for, audio-utils) \ + $(TOPDIR)frameworks/av/services/audiopolicy/common/include \ + $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \ LOCAL_SHARED_LIBRARIES := \ libcutils \ @@ -39,7 +41,8 @@ LOCAL_SHARED_LIBRARIES += \ endif LOCAL_STATIC_LIBRARIES := \ - libmedia_helper + libmedia_helper \ + libaudiopolicycomponents LOCAL_MODULE:= libaudiopolicyservice @@ -53,7 +56,7 @@ ifneq ($(USE_LEGACY_AUDIO_POLICY), 1) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - AudioPolicyManager.cpp + managerdefault/AudioPolicyManager.cpp \ LOCAL_SHARED_LIBRARIES := \ libcutils \ @@ -61,8 +64,15 @@ LOCAL_SHARED_LIBRARIES := \ liblog \ libsoundtrigger +LOCAL_SHARED_LIBRARIES += libaudiopolicyenginedefault + +LOCAL_C_INCLUDES += \ + $(TOPDIR)frameworks/av/services/audiopolicy/common/include \ + $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \ + LOCAL_STATIC_LIBRARIES := \ - libmedia_helper + libmedia_helper \ + libaudiopolicycomponents LOCAL_MODULE:= libaudiopolicymanagerdefault @@ -73,14 +83,26 @@ ifneq ($(USE_CUSTOM_AUDIO_POLICY), 1) include $(CLEAR_VARS) LOCAL_SRC_FILES:= \ - AudioPolicyFactory.cpp + manager/AudioPolicyFactory.cpp LOCAL_SHARED_LIBRARIES := \ libaudiopolicymanagerdefault +LOCAL_STATIC_LIBRARIES := \ + libaudiopolicycomponents + +LOCAL_C_INCLUDES += \ + $(TOPDIR)frameworks/av/services/audiopolicy/common/include \ + $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface \ + LOCAL_MODULE:= libaudiopolicymanager include $(BUILD_SHARED_LIBRARY) endif endif + +####################################################################### +# Recursive call sub-folder Android.mk +# +include $(call all-makefiles-under,$(LOCAL_PATH)) diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h index 4508fa7..48d0e29 100644 --- a/services/audiopolicy/AudioPolicyInterface.h +++ b/services/audiopolicy/AudioPolicyInterface.h @@ -75,7 +75,8 @@ public: // indicate a change in device connection status virtual status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address) = 0; + const char *device_address, + const char *device_name) = 0; // retrieve a device connection status virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address) = 0; @@ -109,6 +110,7 @@ public: audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int selectedDeviceId, const audio_offload_info_t *offloadInfo) = 0; // indicates to the audio policy manager that the output starts being used by corresponding stream. virtual status_t startOutput(audio_io_handle_t output, diff --git a/services/audiopolicy/AudioPolicyManager.h b/services/audiopolicy/AudioPolicyManager.h deleted file mode 100644 index cbdafa6..0000000 --- a/services/audiopolicy/AudioPolicyManager.h +++ /dev/null @@ -1,937 +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); - 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() {} - - 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...). - }; - - 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; - audio_port_handle_t mId; - }; - - class DeviceVector : public SortedVector< sp<DeviceDescriptor> > - { - public: - DeviceVector() : SortedVector(), mDeviceTypes(AUDIO_DEVICE_NONE) {} - - ssize_t add(const sp<DeviceDescriptor>& item); - ssize_t remove(const sp<DeviceDescriptor>& item); - ssize_t indexOf(const sp<DeviceDescriptor>& item) const; - - audio_devices_t types() const { return mDeviceTypes; } - - void loadDevicesFromType(audio_devices_t types); - void loadDevicesFromName(char *name, const DeviceVector& declaredDevices); - - sp<DeviceDescriptor> getDevice(audio_devices_t type, String8 address) const; - DeviceVector getDevicesFromType(audio_devices_t types) const; - sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const; - sp<DeviceDescriptor> getDeviceFromName(const String8& name) const; - 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; - 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); - 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); - sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device, - const char *device_address); - -}; - -}; diff --git a/services/audiopolicy/common/Android.mk b/services/audiopolicy/common/Android.mk new file mode 100644 index 0000000..dcce8e3 --- /dev/null +++ b/services/audiopolicy/common/Android.mk @@ -0,0 +1,9 @@ + +LOCAL_PATH := $(call my-dir) +include $(CLEAR_VARS) + +####################################################################### +# Recursive call sub-folder Android.mk +# +include $(call all-makefiles-under,$(LOCAL_PATH)) + diff --git a/services/audiopolicy/common/include/RoutingStrategy.h b/services/audiopolicy/common/include/RoutingStrategy.h new file mode 100644 index 0000000..d38967e --- /dev/null +++ b/services/audiopolicy/common/include/RoutingStrategy.h @@ -0,0 +1,38 @@ +/* + * 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. + */ + +#pragma once + +namespace android { + +// 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 + +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 +}; + +}; //namespace android diff --git a/services/audiopolicy/common/include/Volume.h b/services/audiopolicy/common/include/Volume.h new file mode 100755 index 0000000..4205589 --- /dev/null +++ b/services/audiopolicy/common/include/Volume.h @@ -0,0 +1,157 @@ +/* + * 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. + */ + +#pragma once + +#include <system/audio.h> +#include <utils/Log.h> +#include <math.h> + +// Absolute min volume in dB (can be represented in single precision normal float value) +#define VOLUME_MIN_DB (-758) + +class VolumeCurvePoint +{ +public: + int mIndex; + float mDBAttenuation; +}; + +class Volume +{ +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 volIndexToDb() + * + * @todo shall become configurable + */ + 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 + }; + + /** + * extract one device relevant for volume control from multiple device selection + * + * @param[in] device for which the volume category is associated + * + * @return subset of device required to limit the number of volume category per device + */ + static audio_devices_t 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; + } + + /** + * returns the category the device belongs to with regard to volume curve management + * + * @param[in] device to check upon the category to whom it belongs to. + * + * @return device category. + */ + static device_category 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 inline float DbToAmpl(float decibels) + { + if (decibels <= VOLUME_MIN_DB) { + return 0.0f; + } + return exp( decibels * 0.115129f); // exp( dB * ln(10) / 20 ) + } + + static inline float AmplToDb(float amplification) + { + if (amplification == 0) { + return VOLUME_MIN_DB; + } + return 20 * log10(amplification); + } + +}; diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h new file mode 100755 index 0000000..a2327ee --- /dev/null +++ b/services/audiopolicy/common/include/policy.h @@ -0,0 +1,84 @@ +/* + * 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. + */ + +#pragma once + +#include <system/audio.h> + +// For mixed output and inputs, the policy will use max mixer sampling rates. +// Do not limit sampling rate otherwise +#define MAX_MIXER_SAMPLING_RATE 48000 + +// For mixed output and inputs, the policy will use max mixer channel count. +// Do not limit channel count otherwise +#define MAX_MIXER_CHANNEL_COUNT 8 + +/** + * A device mask for all audio input devices that are considered "virtual" when evaluating + * active inputs in getActiveInput() + */ +#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER) + + +/** + * A device mask for all audio input and output devices where matching inputs/outputs on device + * type alone is not enough: the address must match too + */ +#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ + AUDIO_DEVICE_OUT_REMOTE_SUBMIX) + +/** + * Check if the state given correspond to an in call state. + * @TODO find a better name for widely call state + * + * @param[in] state to consider + * + * @return true if given state represents a device in a telephony or VoIP call + */ +static inline bool is_state_in_call(int state) +{ + return (state == AUDIO_MODE_IN_CALL) || (state == AUDIO_MODE_IN_COMMUNICATION); +} + +/** + * Check if the input device given is considered as a virtual device. + * + * @param[in] device to consider + * + * @return true if the device is a virtual one, false otherwise. + */ +static bool is_virtual_input_device(audio_devices_t device) +{ + if ((device & AUDIO_DEVICE_BIT_IN) != 0) { + device &= ~AUDIO_DEVICE_BIT_IN; + if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) + return true; + } + return false; +} + +/** + * Check whether the device type is one + * where addresses are used to distinguish between one connected device and another + * + * @param[in] device to consider + * + * @return true if the device needs distinguish on address, false otherwise.. + */ +static bool device_distinguishes_on_address(audio_devices_t device) +{ + return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0); +} diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk new file mode 100644 index 0000000..7c265aa --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/Android.mk @@ -0,0 +1,35 @@ +LOCAL_PATH:= $(call my-dir) + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES:= \ + src/DeviceDescriptor.cpp \ + src/AudioGain.cpp \ + src/StreamDescriptor.cpp \ + src/HwModule.cpp \ + src/IOProfile.cpp \ + src/AudioPort.cpp \ + src/AudioPolicyMix.cpp \ + src/AudioPatch.cpp \ + src/AudioInputDescriptor.cpp \ + src/AudioOutputDescriptor.cpp \ + src/EffectDescriptor.cpp \ + src/ConfigParsingUtils.cpp \ + src/SoundTriggerSession.cpp \ + +LOCAL_SHARED_LIBRARIES := \ + libcutils \ + libutils \ + liblog \ + +LOCAL_C_INCLUDES += \ + $(LOCAL_PATH)/include \ + $(TOPDIR)frameworks/av/services/audiopolicy/common/include \ + $(TOPDIR)frameworks/av/services/audiopolicy + +LOCAL_EXPORT_C_INCLUDE_DIRS := \ + $(LOCAL_PATH)/include + +LOCAL_MODULE := libaudiopolicycomponents + +include $(BUILD_STATIC_LIBRARY) diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h new file mode 100644 index 0000000..21fbf9b --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioGain.h @@ -0,0 +1,40 @@ +/* + * 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. + */ + +#pragma once + +#include <utils/Errors.h> +#include <utils/RefBase.h> +#include <system/audio.h> + +namespace android { + +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; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h new file mode 100644 index 0000000..18bcfdb --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h @@ -0,0 +1,87 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include <utils/Errors.h> +#include <system/audio.h> +#include <utils/SortedVector.h> +#include <utils/KeyedVector.h> + +namespace android { + +class IOProfile; +class AudioMix; + +// 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); + void setIoHandle(audio_io_handle_t ioHandle); + audio_port_handle_t getId() const; + audio_module_handle_t getModuleHandle() const; + + status_t dump(int fd); + + 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; + +private: + audio_port_handle_t mId; +}; + +class AudioInputCollection : + public DefaultKeyedVector< audio_io_handle_t, sp<AudioInputDescriptor> > +{ +public: + bool isSourceActive(audio_source_t source) const; + + sp<AudioInputDescriptor> getInputFromId(audio_port_handle_t id) const; + + uint32_t activeInputsCount() const; + + /** + * 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); + + audio_devices_t getSupportedDevices(audio_io_handle_t handle) const; + + status_t dump(int fd) const; +}; + + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h new file mode 100644 index 0000000..f1aee46 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h @@ -0,0 +1,162 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include <RoutingStrategy.h> +#include <utils/Errors.h> +#include <utils/Timers.h> +#include <utils/KeyedVector.h> +#include <system/audio.h> + +namespace android { + +class IOProfile; +class AudioMix; +class AudioPolicyClientInterface; + +// 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<AudioPort>& port, + AudioPolicyClientInterface *clientInterface); + virtual ~AudioOutputDescriptor() {} + + status_t dump(int fd); + void log(const char* indent); + + audio_port_handle_t getId() const; + virtual audio_devices_t device() const; + virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); + virtual audio_devices_t supportedDevices(); + virtual bool isDuplicated() const { return false; } + virtual uint32_t latency() { return 0; } + virtual bool isFixedVolume(audio_devices_t device); + virtual sp<AudioOutputDescriptor> subOutput1() { return 0; } + virtual sp<AudioOutputDescriptor> subOutput2() { return 0; } + virtual bool setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force); + virtual void changeRefCount(audio_stream_type_t stream, int delta); + + bool isActive(uint32_t inPastMs = 0) const; + bool isStreamActive(audio_stream_type_t stream, + 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 mPort; } + virtual void toAudioPort(struct audio_port *port) const; + + audio_module_handle_t getModuleHandle() const; + + sp<AudioPort> mPort; + audio_devices_t mDevice; // current device this output is routed to + audio_patch_handle_t mPatchHandle; + uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output + nsecs_t mStopTime[AUDIO_STREAM_CNT]; + float mCurVolume[AUDIO_STREAM_CNT]; // current stream volume in dB + int mMuteCount[AUDIO_STREAM_CNT]; // mute request counter + bool mStrategyMutedByDevice[NUM_STRATEGIES]; // strategies muted because of incompatible + // device selection. See checkDeviceMuteStrategies() + AudioPolicyClientInterface *mClientInterface; + +protected: + audio_port_handle_t mId; +}; + +// Audio output driven by a software mixer in audio flinger. +class SwAudioOutputDescriptor: public AudioOutputDescriptor +{ +public: + SwAudioOutputDescriptor(const sp<IOProfile>& profile, + AudioPolicyClientInterface *clientInterface); + virtual ~SwAudioOutputDescriptor() {} + + status_t dump(int fd); + + void setIoHandle(audio_io_handle_t ioHandle); + + virtual audio_devices_t device() const; + virtual bool sharesHwModuleWith(const sp<AudioOutputDescriptor> outputDesc); + virtual audio_devices_t supportedDevices(); + virtual uint32_t latency(); + virtual bool isDuplicated() const { return (mOutput1 != NULL && mOutput2 != NULL); } + virtual bool isFixedVolume(audio_devices_t device); + virtual sp<AudioOutputDescriptor> subOutput1() { return mOutput1; } + virtual sp<AudioOutputDescriptor> subOutput2() { return mOutput2; } + virtual void changeRefCount(audio_stream_type_t stream, int delta); + virtual bool setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force); + + virtual void toAudioPortConfig(struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig = NULL) const; + virtual void toAudioPort(struct audio_port *port) const; + + const sp<IOProfile> mProfile; // I/O profile this output derives from + audio_io_handle_t mIoHandle; // output handle + uint32_t mLatency; // + audio_output_flags_t mFlags; // + AudioMix *mPolicyMix; // non NULL when used by a dynamic policy + sp<SwAudioOutputDescriptor> mOutput1; // used by duplicated outputs: first output + sp<SwAudioOutputDescriptor> mOutput2; // used by duplicated outputs: second output + uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only) +}; + +class SwAudioOutputCollection : + public DefaultKeyedVector< audio_io_handle_t, sp<SwAudioOutputDescriptor> > +{ +public: + 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. + */ + bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; + + /** + * returns the A2DP output handle if it is open or 0 otherwise + */ + audio_io_handle_t getA2dpOutput() const; + + sp<SwAudioOutputDescriptor> getOutputFromId(audio_port_handle_t id) const; + + sp<SwAudioOutputDescriptor> getPrimaryOutput() const; + + /** + * return true if any output is playing anything besides the stream to ignore + */ + bool isAnyOutputActive(audio_stream_type_t streamToIgnore) const; + + audio_devices_t getSupportedDevices(audio_io_handle_t handle) const; + + status_t dump(int fd) const; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h new file mode 100644 index 0000000..385f257 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPatch.h @@ -0,0 +1,55 @@ +/* + * 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. + */ + +#pragma once + +#include <system/audio.h> +#include <utils/Errors.h> +#include <utils/RefBase.h> +#include <utils/Errors.h> +#include <utils/KeyedVector.h> + +namespace android { + +class AudioPatch : public RefBase +{ +public: + AudioPatch(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; + +private: + static volatile int32_t mNextUniqueId; +}; + +class AudioPatchCollection : public DefaultKeyedVector<audio_patch_handle_t, sp<AudioPatch> > +{ +public: + status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch); + + status_t removeAudioPatch(audio_patch_handle_t handle); + + status_t listAudioPatches(unsigned int *num_patches, struct audio_patch *patches) const; + + status_t dump(int fd) const; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h new file mode 100644 index 0000000..d51f4e1 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyMix.h @@ -0,0 +1,81 @@ +/* + * 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. + */ + +#pragma once + +#include <utils/RefBase.h> +#include <media/AudioPolicy.h> +#include <utils/KeyedVector.h> +#include <hardware/audio.h> +#include <utils/String8.h> + +namespace android { + +class SwAudioOutputDescriptor; + +/** + * custom mix entry in mPolicyMixes + */ +class AudioPolicyMix : public RefBase { +public: + AudioPolicyMix() {} + + const sp<SwAudioOutputDescriptor> &getOutput() const; + + void setOutput(sp<SwAudioOutputDescriptor> &output); + + void clearOutput(); + + android::AudioMix *getMix(); + + void setMix(AudioMix &mix); + +private: + AudioMix mMix; // Audio policy mix descriptor + sp<SwAudioOutputDescriptor> mOutput; // Corresponding output stream +}; + + +class AudioPolicyMixCollection : public DefaultKeyedVector<String8, sp<AudioPolicyMix> > +{ +public: + status_t getAudioPolicyMix(String8 address, sp<AudioPolicyMix> &policyMix) const; + + status_t registerMix(String8 address, AudioMix mix); + + status_t unregisterMix(String8 address); + + void closeOutput(sp<SwAudioOutputDescriptor> &desc); + + /** + * Try to find an output descriptor for the given attributes. + * + * @param[in] attributes to consider fowr the research of output descriptor. + * @param[out] desc to return if an output could be found. + * + * @return NO_ERROR if an output was found for the given attribute (in this case, the + * descriptor output param is initialized), error code otherwise. + */ + status_t getOutputForAttr(audio_attributes_t attributes, sp<SwAudioOutputDescriptor> &desc); + + audio_devices_t getDeviceAndMixForInputSource(audio_source_t inputSource, + audio_devices_t availableDeviceTypes, + AudioMix **policyMix); + + status_t getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix); +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h new file mode 100644 index 0000000..dea1b8a --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h @@ -0,0 +1,119 @@ +/* + * 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. + */ + +#pragma once + +#include <utils/String8.h> +#include <utils/Vector.h> +#include <utils/RefBase.h> +#include <utils/Errors.h> +#include <system/audio.h> +#include <cutils/config_utils.h> + +namespace android { + +class HwModule; +class AudioGain; + +class AudioPort : public virtual RefBase +{ +public: + AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role); + virtual ~AudioPort() {} + + virtual void attach(const sp<HwModule>& module); + bool isAttached() { return mModule != 0; } + + static audio_port_handle_t getNextUniqueId(); + + 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); + + audio_module_handle_t getModuleHandle() const; + uint32_t getModuleVersion() const; + const char *getModuleName() const; + + void dump(int fd, int spaces) const; + void log(const char* indent) 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...). + +private: + static volatile int32_t mNextUniqueId; +}; + +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; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h new file mode 100644 index 0000000..f8c4d08 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/ConfigParsingUtils.h @@ -0,0 +1,254 @@ +/* + * 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. + */ + +#pragma once + +#include "DeviceDescriptor.h" +#include "HwModule.h" +#include "audio_policy_conf.h" +#include <system/audio.h> +#include <utils/Log.h> +#include <utils/Vector.h> +#include <utils/SortedVector.h> +#include <cutils/config_utils.h> +#include <utils/RefBase.h> +#include <system/audio_policy.h> + +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 NAME_TO_ENUM(name, value) { name, value } +#ifndef ARRAY_SIZE +#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) +#endif + +const StringToEnum sDeviceTypeToEnumTable[] = { + 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 sDeviceNameToEnumTable[] = { + NAME_TO_ENUM("Earpiece", AUDIO_DEVICE_OUT_EARPIECE), + NAME_TO_ENUM("Speaker", AUDIO_DEVICE_OUT_SPEAKER), + NAME_TO_ENUM("Speaker Protected", AUDIO_DEVICE_OUT_SPEAKER_SAFE), + NAME_TO_ENUM("Wired Headset", AUDIO_DEVICE_OUT_WIRED_HEADSET), + NAME_TO_ENUM("Wired Headphones", AUDIO_DEVICE_OUT_WIRED_HEADPHONE), + NAME_TO_ENUM("BT SCO", AUDIO_DEVICE_OUT_BLUETOOTH_SCO), + NAME_TO_ENUM("BT SCO Headset", AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET), + NAME_TO_ENUM("BT SCO Car Kit", AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_SCO), + NAME_TO_ENUM("BT A2DP Out", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP), + NAME_TO_ENUM("BT A2DP Headphones", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES), + NAME_TO_ENUM("BT A2DP Speaker", AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_A2DP), + NAME_TO_ENUM("HDMI Out", AUDIO_DEVICE_OUT_AUX_DIGITAL), + NAME_TO_ENUM("HDMI Out", AUDIO_DEVICE_OUT_HDMI), + NAME_TO_ENUM("Analog Dock Out", AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET), + NAME_TO_ENUM("Digital Dock Out", AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET), + NAME_TO_ENUM("USB Host Out", AUDIO_DEVICE_OUT_USB_ACCESSORY), + NAME_TO_ENUM("USB Device Out", AUDIO_DEVICE_OUT_USB_DEVICE), + NAME_TO_ENUM("", AUDIO_DEVICE_OUT_ALL_USB), + NAME_TO_ENUM("Reroute Submix Out", AUDIO_DEVICE_OUT_REMOTE_SUBMIX), + NAME_TO_ENUM("Telephony Tx", AUDIO_DEVICE_OUT_TELEPHONY_TX), + NAME_TO_ENUM("Line Out", AUDIO_DEVICE_OUT_LINE), + NAME_TO_ENUM("HDMI ARC Out", AUDIO_DEVICE_OUT_HDMI_ARC), + NAME_TO_ENUM("S/PDIF Out", AUDIO_DEVICE_OUT_SPDIF), + NAME_TO_ENUM("FM transceiver Out", AUDIO_DEVICE_OUT_FM), + NAME_TO_ENUM("Aux Line Out", AUDIO_DEVICE_OUT_AUX_LINE), + NAME_TO_ENUM("Ambient Mic", AUDIO_DEVICE_IN_AMBIENT), + NAME_TO_ENUM("Built-In Mic", AUDIO_DEVICE_IN_BUILTIN_MIC), + NAME_TO_ENUM("BT SCO Headset Mic", AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET), + NAME_TO_ENUM("", AUDIO_DEVICE_IN_ALL_SCO), + NAME_TO_ENUM("Wired Headset Mic", AUDIO_DEVICE_IN_WIRED_HEADSET), + NAME_TO_ENUM("HDMI In", AUDIO_DEVICE_IN_AUX_DIGITAL), + NAME_TO_ENUM("HDMI In", AUDIO_DEVICE_IN_HDMI), + NAME_TO_ENUM("Telephony Rx", AUDIO_DEVICE_IN_TELEPHONY_RX), + NAME_TO_ENUM("Telephony Rx", AUDIO_DEVICE_IN_VOICE_CALL), + NAME_TO_ENUM("Built-In Back Mic", AUDIO_DEVICE_IN_BACK_MIC), + NAME_TO_ENUM("Reroute Submix In", AUDIO_DEVICE_IN_REMOTE_SUBMIX), + NAME_TO_ENUM("Analog Dock In", AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET), + NAME_TO_ENUM("Digital Dock In", AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET), + NAME_TO_ENUM("USB Host In", AUDIO_DEVICE_IN_USB_ACCESSORY), + NAME_TO_ENUM("USB Device In", AUDIO_DEVICE_IN_USB_DEVICE), + NAME_TO_ENUM("FM Tuner In", AUDIO_DEVICE_IN_FM_TUNER), + NAME_TO_ENUM("TV Tuner In", AUDIO_DEVICE_IN_TV_TUNER), + NAME_TO_ENUM("Line In", AUDIO_DEVICE_IN_LINE), + NAME_TO_ENUM("S/PDIF In", AUDIO_DEVICE_IN_SPDIF), + NAME_TO_ENUM("BT A2DP In", AUDIO_DEVICE_IN_BLUETOOTH_A2DP), + NAME_TO_ENUM("Loopback In", 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), + STRING_TO_ENUM(AUDIO_FORMAT_DTS), + STRING_TO_ENUM(AUDIO_FORMAT_DTS_HD), +}; + +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); + + static void loadHwModules(cnode *root, HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled); + + static void loadGlobalConfig(cnode *root, const sp<HwModule>& module, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled); + + static status_t loadAudioPolicyConfig(const char *path, + HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled); + +private: + static void loadHwModule(cnode *root, HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled); +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h new file mode 100644 index 0000000..aa37eec --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h @@ -0,0 +1,95 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include <utils/Errors.h> +#include <utils/String8.h> +#include <utils/SortedVector.h> +#include <cutils/config_utils.h> +#include <system/audio.h> +#include <system/audio_policy.h> + +namespace android { + +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 attach(const sp<HwModule>& module); + virtual void loadGains(cnode *root); + virtual void toAudioPort(struct audio_port *port) const; + + audio_port_handle_t getId() const; + audio_devices_t type() const { return mDeviceType; } + status_t dump(int fd, int spaces, int index) const; + void log() const; + + String8 mAddress; + + static String8 emptyNameStr; + +private: + audio_devices_t mDeviceType; + audio_port_handle_t mId; + +friend class DeviceVector; +}; + +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; + + audio_devices_t getDevicesFromHwModule(audio_module_handle_t moduleHandle) const; + + audio_policy_dev_state_t getDeviceConnectionState(const sp<DeviceDescriptor> &devDesc) const; + + status_t dump(int fd, const String8 &direction) const; + +private: + void refreshTypes(); + audio_devices_t mDeviceTypes; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h new file mode 100644 index 0000000..c9783a1 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/EffectDescriptor.h @@ -0,0 +1,71 @@ +/* + * 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. + */ + +#pragma once + +#include <RoutingStrategy.h> +#include <hardware/audio_effect.h> +#include <utils/KeyedVector.h> +#include <utils/RefBase.h> +#include <utils/Errors.h> + +namespace android { + + +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 +}; + +class EffectDescriptorCollection : public KeyedVector<int, sp<EffectDescriptor> > +{ +public: + EffectDescriptorCollection(); + + status_t registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, + uint32_t strategy, int session, int id); + status_t unregisterEffect(int id); + status_t setEffectEnabled(int id, bool enabled); + uint32_t getMaxEffectsCpuLoad() const; + uint32_t getMaxEffectsMemory() const; + bool isNonOffloadableEffectEnabled(); + + status_t dump(int fd); + +private: + status_t setEffectEnabled(const sp<EffectDescriptor> &effectDesc, bool enabled); + + uint32_t mTotalEffectsCpuLoad; // current CPU load used by effects + uint32_t mTotalEffectsMemory; // current memory used by effects + + /** + * 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; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h new file mode 100644 index 0000000..92c3ea2 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h @@ -0,0 +1,74 @@ +/* + * 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. + */ + +#pragma once + +#include "DeviceDescriptor.h" +#include <utils/RefBase.h> +#include <utils/String8.h> +#include <utils/Errors.h> +#include <utils/Vector.h> +#include <system/audio.h> +#include <cutils/config_utils.h> + +namespace android { + +class IOProfile; + +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); + + audio_module_handle_t getHandle() const { return mHandle; } + + 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 +}; + +class HwModuleCollection : public Vector< sp<HwModule> > +{ +public: + sp<HwModule> getModuleFromName(const char *name) const; + + sp <HwModule> getModuleForDevice(audio_devices_t device) const; + + sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device, + const char *device_address, + const char *device_name) const; + + status_t dump(int fd) const; +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h new file mode 100644 index 0000000..022257e --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h @@ -0,0 +1,58 @@ +/* + * 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. + */ + +#pragma once + +#include "AudioPort.h" +#include "DeviceDescriptor.h" +#include <utils/String8.h> +#include <system/audio.h> + +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); + 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/common/managerdefinitions/include/SoundTriggerSession.h b/services/audiopolicy/common/managerdefinitions/include/SoundTriggerSession.h new file mode 100644 index 0000000..420e6d7 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/SoundTriggerSession.h @@ -0,0 +1,33 @@ +/* + * 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. + */ + +#pragma once + +#include <system/audio.h> +#include <utils/Errors.h> +#include <utils/KeyedVector.h> + +namespace android { + +class SoundTriggerSessionCollection : public DefaultKeyedVector<audio_session_t, audio_io_handle_t> +{ +public: + status_t releaseSession(audio_session_t session); + + status_t acquireSession(audio_session_t session, audio_io_handle_t ioHandle); +}; + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h new file mode 100644 index 0000000..84db5ab --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/include/StreamDescriptor.h @@ -0,0 +1,85 @@ +/* + * 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. + */ + +#pragma once + +#include <Volume.h> +#include <utils/KeyedVector.h> +#include <utils/StrongPointer.h> +#include <utils/SortedVector.h> +#include <hardware/audio.h> + +namespace android { + +// stream descriptor used for volume control +class StreamDescriptor +{ +public: + StreamDescriptor(); + + int getVolumeIndex(audio_devices_t device) const; + bool canBeMuted() const { return mCanBeMuted; } + void clearCurrentVolumeIndex(); + void addCurrentVolumeIndex(audio_devices_t device, int index); + int getVolumeIndexMin() const { return mIndexMin; } + int getVolumeIndexMax() const { return mIndexMax; } + void setVolumeIndexMin(int volIndexMin); + void setVolumeIndexMax(int volIndexMax); + + void dump(int fd) const; + + void setVolumeCurvePoint(Volume::device_category deviceCategory, const VolumeCurvePoint *point); + const VolumeCurvePoint *getVolumeCurvePoint(Volume::device_category deviceCategory) const + { + return mVolumeCurve[deviceCategory]; + } + +private: + const VolumeCurvePoint *mVolumeCurve[Volume::DEVICE_CATEGORY_CNT]; + KeyedVector<audio_devices_t, int> mIndexCur; /**< current volume index per device. */ + int mIndexMin; /**< min volume index. */ + int mIndexMax; /**< max volume index. */ + bool mCanBeMuted; /**< true is the stream can be muted. */ +}; + +/** + * stream descriptors collection for volume control + */ +class StreamDescriptorCollection : public DefaultKeyedVector<audio_stream_type_t, StreamDescriptor> +{ +public: + StreamDescriptorCollection(); + + void clearCurrentVolumeIndex(audio_stream_type_t stream); + void addCurrentVolumeIndex(audio_stream_type_t stream, audio_devices_t device, int index); + + bool canBeMuted(audio_stream_type_t stream); + + status_t dump(int fd) const; + + void setVolumeCurvePoint(audio_stream_type_t stream, + Volume::device_category deviceCategory, + const VolumeCurvePoint *point); + + const VolumeCurvePoint *getVolumeCurvePoint(audio_stream_type_t stream, + Volume::device_category deviceCategory) const; + + void setVolumeIndexMin(audio_stream_type_t stream,int volIndexMin); + void setVolumeIndexMax(audio_stream_type_t stream,int volIndexMax); + +}; + +}; // namespace android diff --git a/services/audiopolicy/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h index 2535a67..a393e3b 100644 --- a/services/audiopolicy/audio_policy_conf.h +++ b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h @@ -14,9 +14,7 @@ * limitations under the License. */ - -#ifndef ANDROID_AUDIO_POLICY_CONF_H -#define ANDROID_AUDIO_POLICY_CONF_H +#pragma once ///////////////////////////////////////////////// @@ -53,9 +51,9 @@ // "formats" in outputs descriptors indicating that supported // values should be queried after opening the output. -#define DEVICES_TAG "devices" -#define DEVICE_TYPE "type" -#define DEVICE_ADDRESS "address" +#define APM_DEVICES_TAG "devices" +#define APM_DEVICE_TYPE "type" +#define APM_DEVICE_ADDRESS "address" #define MIXERS_TAG "mixers" #define MIXER_TYPE "type" @@ -71,7 +69,3 @@ #define GAIN_STEP_VALUE "step_value_mB" #define GAIN_MIN_RAMP_MS "min_ramp_ms" #define GAIN_MAX_RAMP_MS "max_ramp_ms" - - - -#endif // ANDROID_AUDIO_POLICY_CONF_H diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp new file mode 100644 index 0000000..fc7b0cc --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp @@ -0,0 +1,130 @@ +/* + * 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::AudioGain" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include "AudioGain.h" +#include "StreamDescriptor.h" +#include <utils/Log.h> +#include <utils/String8.h> + +#include <math.h> + +namespace android { + +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()); +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp new file mode 100644 index 0000000..937160b --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp @@ -0,0 +1,195 @@ +/* + * 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 "AudioInputDescriptor.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "HwModule.h" +#include <media/AudioPolicy.h> +#include <policy.h> + +namespace android { + +AudioInputDescriptor::AudioInputDescriptor(const sp<IOProfile>& profile) + : mIoHandle(0), + mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL), mPatchHandle(0), mRefCount(0), + mInputSource(AUDIO_SOURCE_DEFAULT), mProfile(profile), mIsSoundTrigger(false), mId(0) +{ + if (profile != NULL) { + mSamplingRate = profile->pickSamplingRate(); + mFormat = profile->pickFormat(); + mChannelMask = profile->pickChannelMask(); + if (profile->mGains.size() > 0) { + profile->mGains[0]->getDefaultConfig(&mGain); + } + } +} + +void AudioInputDescriptor::setIoHandle(audio_io_handle_t ioHandle) +{ + mId = AudioPort::getNextUniqueId(); + mIoHandle = ioHandle; +} + +audio_module_handle_t AudioInputDescriptor::getModuleHandle() const +{ + if (mProfile == 0) { + return 0; + } + return mProfile->getModuleHandle(); +} + +audio_port_handle_t AudioInputDescriptor::getId() const +{ + return mId; +} + +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 = getModuleHandle(); + 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 = getModuleHandle(); + 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", getId()); + 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; +} + +bool AudioInputCollection::isSourceActive(audio_source_t source) const +{ + for (size_t i = 0; i < size(); i++) { + const sp<AudioInputDescriptor> inputDescriptor = valueAt(i); + if (inputDescriptor->mRefCount == 0) { + continue; + } + if (inputDescriptor->mInputSource == (int)source) { + return true; + } + } + return false; +} + +sp<AudioInputDescriptor> AudioInputCollection::getInputFromId(audio_port_handle_t id) const +{ + sp<AudioInputDescriptor> inputDesc = NULL; + for (size_t i = 0; i < size(); i++) { + inputDesc = valueAt(i); + if (inputDesc->getId() == id) { + break; + } + } + return inputDesc; +} + +uint32_t AudioInputCollection::activeInputsCount() const +{ + uint32_t count = 0; + for (size_t i = 0; i < size(); i++) { + const sp<AudioInputDescriptor> desc = valueAt(i); + if (desc->mRefCount > 0) { + count++; + } + } + return count; +} + +audio_io_handle_t AudioInputCollection::getActiveInput(bool ignoreVirtualInputs) +{ + for (size_t i = 0; i < size(); i++) { + const sp<AudioInputDescriptor> input_descriptor = valueAt(i); + if ((input_descriptor->mRefCount > 0) + && (!ignoreVirtualInputs || !is_virtual_input_device(input_descriptor->mDevice))) { + return keyAt(i); + } + } + return 0; +} + +audio_devices_t AudioInputCollection::getSupportedDevices(audio_io_handle_t handle) const +{ + sp<AudioInputDescriptor> inputDesc = valueFor(handle); + audio_devices_t devices = inputDesc->mProfile->mSupportedDevices.types(); + return devices; +} + +status_t AudioInputCollection::dump(int fd) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\nInputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + snprintf(buffer, SIZE, "- Input %d dump:\n", keyAt(i)); + write(fd, buffer, strlen(buffer)); + valueAt(i)->dump(fd); + } + + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp new file mode 100644 index 0000000..596aa1d --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp @@ -0,0 +1,474 @@ +/* + * 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 <AudioPolicyInterface.h> +#include "AudioOutputDescriptor.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "Volume.h" +#include "HwModule.h" +#include <media/AudioPolicy.h> + +// A device mask for all audio output devices that are considered "remote" when evaluating +// active output devices in isStreamActiveRemotely() +#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX + +namespace android { + +AudioOutputDescriptor::AudioOutputDescriptor(const sp<AudioPort>& port, + AudioPolicyClientInterface *clientInterface) + : mPort(port), mDevice(AUDIO_DEVICE_NONE), + mPatchHandle(0), mClientInterface(clientInterface), mId(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 (port != NULL) { + mSamplingRate = port->pickSamplingRate(); + mFormat = port->pickFormat(); + mChannelMask = port->pickChannelMask(); + if (port->mGains.size() > 0) { + port->mGains[0]->getDefaultConfig(&mGain); + } + } +} + +audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const +{ + return mPort->getModuleHandle(); +} + +audio_port_handle_t AudioOutputDescriptor::getId() const +{ + return mId; +} + +audio_devices_t AudioOutputDescriptor::device() const +{ + return mDevice; +} + +audio_devices_t AudioOutputDescriptor::supportedDevices() +{ + return mDevice; +} + +bool AudioOutputDescriptor::sharesHwModuleWith( + const sp<AudioOutputDescriptor> outputDesc) +{ + if (outputDesc->isDuplicated()) { + return sharesHwModuleWith(outputDesc->subOutput1()) || + sharesHwModuleWith(outputDesc->subOutput2()); + } else { + return (getModuleHandle() == outputDesc->getModuleHandle()); + } +} + +void AudioOutputDescriptor::changeRefCount(audio_stream_type_t stream, + int 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]); +} + +bool AudioOutputDescriptor::isActive(uint32_t inPastMs) const +{ + nsecs_t sysTime = 0; + if (inPastMs != 0) { + sysTime = systemTime(); + } + for (int i = 0; i < (int)AUDIO_STREAM_CNT; i++) { + if (i == AUDIO_STREAM_PATCH) { + continue; + } + if (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; +} + + +bool AudioOutputDescriptor::isFixedVolume(audio_devices_t device __unused) +{ + return false; +} + +bool AudioOutputDescriptor::setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device __unused, + uint32_t delayMs, + bool force) +{ + // We actually change the volume if: + // - the float value returned by computeVolume() changed + // - the force flag is set + if (volume != mCurVolume[stream] || force) { + ALOGV("setVolume() for stream %d, volume %f, delay %d", stream, volume, delayMs); + mCurVolume[stream] = volume; + return true; + } + return false; +} + +void AudioOutputDescriptor::toAudioPortConfig( + struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + 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 = getModuleHandle(); + dstConfig->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; +} + +void AudioOutputDescriptor::toAudioPort( + struct audio_port *port) const +{ + mPort->toAudioPort(port); + port->id = mId; + port->ext.mix.hw_module = getModuleHandle(); +} + +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, " 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; +} + +void AudioOutputDescriptor::log(const char* indent) +{ + ALOGI("%sID: %d,0x%X, [rt:%d fmt:0x%X ch:0x%X]", + indent, mId, mId, mSamplingRate, mFormat, mChannelMask); +} + +// SwAudioOutputDescriptor implementation +SwAudioOutputDescriptor::SwAudioOutputDescriptor( + const sp<IOProfile>& profile, AudioPolicyClientInterface *clientInterface) + : AudioOutputDescriptor(profile, clientInterface), + mProfile(profile), mIoHandle(0), mLatency(0), + mFlags((audio_output_flags_t)0), mPolicyMix(NULL), + mOutput1(0), mOutput2(0), mDirectOpenCount(0) +{ + if (profile != NULL) { + mFlags = (audio_output_flags_t)profile->mFlags; + } +} + +void SwAudioOutputDescriptor::setIoHandle(audio_io_handle_t ioHandle) +{ + mId = AudioPort::getNextUniqueId(); + mIoHandle = ioHandle; +} + + +status_t SwAudioOutputDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " Latency: %d\n", mLatency); + result.append(buffer); + snprintf(buffer, SIZE, " Flags %08x\n", mFlags); + result.append(buffer); + write(fd, result.string(), result.size()); + + AudioOutputDescriptor::dump(fd); + + return NO_ERROR; +} + +audio_devices_t SwAudioOutputDescriptor::device() const +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->mDevice | mOutput2->mDevice); + } else { + return mDevice; + } +} + +bool SwAudioOutputDescriptor::sharesHwModuleWith( + const sp<AudioOutputDescriptor> outputDesc) +{ + if (isDuplicated()) { + return mOutput1->sharesHwModuleWith(outputDesc) || mOutput2->sharesHwModuleWith(outputDesc); + } else if (outputDesc->isDuplicated()){ + return sharesHwModuleWith(outputDesc->subOutput1()) || + sharesHwModuleWith(outputDesc->subOutput2()); + } else { + return AudioOutputDescriptor::sharesHwModuleWith(outputDesc); + } +} + +audio_devices_t SwAudioOutputDescriptor::supportedDevices() +{ + if (isDuplicated()) { + return (audio_devices_t)(mOutput1->supportedDevices() | mOutput2->supportedDevices()); + } else { + return mProfile->mSupportedDevices.types() ; + } +} + +uint32_t SwAudioOutputDescriptor::latency() +{ + if (isDuplicated()) { + return (mOutput1->mLatency > mOutput2->mLatency) ? mOutput1->mLatency : mOutput2->mLatency; + } else { + return mLatency; + } +} + +void SwAudioOutputDescriptor::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); + } + AudioOutputDescriptor::changeRefCount(stream, delta); +} + + +bool SwAudioOutputDescriptor::isFixedVolume(audio_devices_t device) +{ + // unit gain if rerouting to external policy + if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { + if (mPolicyMix != NULL) { + ALOGV("max gain when rerouting for output=%d", mIoHandle); + return true; + } + } + return false; +} + +void SwAudioOutputDescriptor::toAudioPortConfig( + struct audio_port_config *dstConfig, + const struct audio_port_config *srcConfig) const +{ + + ALOG_ASSERT(!isDuplicated(), "toAudioPortConfig() called on duplicated output %d", mIoHandle); + AudioOutputDescriptor::toAudioPortConfig(dstConfig, srcConfig); + + dstConfig->ext.mix.handle = mIoHandle; +} + +void SwAudioOutputDescriptor::toAudioPort( + struct audio_port *port) const +{ + ALOG_ASSERT(!isDuplicated(), "toAudioPort() called on duplicated output %d", mIoHandle); + + AudioOutputDescriptor::toAudioPort(port); + + toAudioPortConfig(&port->active_config); + port->ext.mix.handle = mIoHandle; + port->ext.mix.latency_class = + mFlags & AUDIO_OUTPUT_FLAG_FAST ? AUDIO_LATENCY_LOW : AUDIO_LATENCY_NORMAL; +} + +bool SwAudioOutputDescriptor::setVolume(float volume, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t delayMs, + bool force) +{ + bool changed = AudioOutputDescriptor::setVolume(volume, stream, device, delayMs, force); + + if (changed) { + // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is + // enabled + float volume = Volume::DbToAmpl(mCurVolume[stream]); + if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { + mClientInterface->setStreamVolume( + AUDIO_STREAM_VOICE_CALL, volume, mIoHandle, delayMs); + } + mClientInterface->setStreamVolume(stream, volume, mIoHandle, delayMs); + } + return changed; +} + +// SwAudioOutputCollection implementation + +bool SwAudioOutputCollection::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < this->size(); i++) { + const sp<SwAudioOutputDescriptor> outputDesc = this->valueAt(i); + if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +bool SwAudioOutputCollection::isStreamActiveRemotely(audio_stream_type_t stream, + uint32_t inPastMs) const +{ + nsecs_t sysTime = systemTime(); + for (size_t i = 0; i < size(); i++) { + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); + if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && + outputDesc->isStreamActive(stream, inPastMs, sysTime)) { + // do not consider re routing (when the output is going to a dynamic policy) + // as "remote playback" + if (outputDesc->mPolicyMix == NULL) { + return true; + } + } + } + return false; +} + +audio_io_handle_t SwAudioOutputCollection::getA2dpOutput() const +{ + for (size_t i = 0; i < size(); i++) { + sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); + if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { + return this->keyAt(i); + } + } + return 0; +} + +sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getPrimaryOutput() const +{ + for (size_t i = 0; i < size(); i++) { + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); + if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { + return outputDesc; + } + } + return NULL; +} + +sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputFromId(audio_port_handle_t id) const +{ + sp<SwAudioOutputDescriptor> outputDesc = NULL; + for (size_t i = 0; i < size(); i++) { + outputDesc = valueAt(i); + if (outputDesc->getId() == id) { + break; + } + } + return outputDesc; +} + +bool SwAudioOutputCollection::isAnyOutputActive(audio_stream_type_t streamToIgnore) const +{ + for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { + if (s == (size_t) streamToIgnore) { + continue; + } + for (size_t i = 0; i < size(); i++) { + const sp<SwAudioOutputDescriptor> outputDesc = valueAt(i); + if (outputDesc->mRefCount[s] != 0) { + return true; + } + } + } + return false; +} + +audio_devices_t SwAudioOutputCollection::getSupportedDevices(audio_io_handle_t handle) const +{ + sp<SwAudioOutputDescriptor> outputDesc = valueFor(handle); + audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); + return devices; +} + + +status_t SwAudioOutputCollection::dump(int fd) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\nOutputs dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + snprintf(buffer, SIZE, "- Output %d dump:\n", keyAt(i)); + write(fd, buffer, strlen(buffer)); + valueAt(i)->dump(fd); + } + + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp new file mode 100644 index 0000000..a06d867 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp @@ -0,0 +1,154 @@ +/* + * 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::AudioPatch" +//#define LOG_NDEBUG 0 + +#include "AudioPatch.h" +#include "AudioGain.h" +#include "ConfigParsingUtils.h" +#include <cutils/log.h> +#include <utils/String8.h> + +namespace android { + +int32_t volatile AudioPatch::mNextUniqueId = 1; + +AudioPatch::AudioPatch(const struct audio_patch *patch, uid_t uid) : + mHandle(static_cast<audio_patch_handle_t>(android_atomic_inc(&mNextUniqueId))), + 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(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + 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(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + 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; +} + +status_t AudioPatchCollection::addAudioPatch(audio_patch_handle_t handle, + const sp<AudioPatch>& patch) +{ + ssize_t index = indexOfKey(handle); + + if (index >= 0) { + ALOGW("addAudioPatch() patch %d already in", handle); + return ALREADY_EXISTS; + } + add(handle, patch); + ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d" + "sink handle %d", + handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks, + patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id); + return NO_ERROR; +} + +status_t AudioPatchCollection::removeAudioPatch(audio_patch_handle_t handle) +{ + ssize_t index = indexOfKey(handle); + + if (index < 0) { + ALOGW("removeAudioPatch() patch %d not in", handle); + return ALREADY_EXISTS; + } + ALOGV("removeAudioPatch() handle %d af handle %d", handle, valueAt(index)->mAfPatchHandle); + removeItemsAt(index); + return NO_ERROR; +} + +status_t AudioPatchCollection::listAudioPatches(unsigned int *num_patches, + struct audio_patch *patches) const +{ + if (num_patches == NULL || (*num_patches != 0 && patches == NULL)) { + return BAD_VALUE; + } + ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu", + *num_patches, patches, size()); + if (patches == NULL) { + *num_patches = 0; + } + + size_t patchesWritten = 0; + size_t patchesMax = *num_patches; + for (size_t i = 0; i < size() && patchesWritten < patchesMax; i++) { + const sp<AudioPatch> patch = valueAt(i); + patches[patchesWritten] = patch->mPatch; + patches[patchesWritten++].id = patch->mHandle; + ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d", + i, patch->mPatch.num_sources, patch->mPatch.num_sinks); + } + *num_patches = size(); + + ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches); + return NO_ERROR; +} + +status_t AudioPatchCollection::dump(int fd) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + snprintf(buffer, SIZE, "\nAudio Patches:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + valueAt(i)->dump(fd, 2, i); + } + return NO_ERROR; +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp new file mode 100644 index 0000000..77fc0b9 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp @@ -0,0 +1,193 @@ +/* + * 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::AudioPolicyMix" +//#define LOG_NDEBUG 0 + +#include "AudioPolicyMix.h" +#include "HwModule.h" +#include "AudioPort.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include <AudioOutputDescriptor.h> + +namespace android { + +void AudioPolicyMix::setOutput(sp<SwAudioOutputDescriptor> &output) +{ + mOutput = output; +} + +const sp<SwAudioOutputDescriptor> &AudioPolicyMix::getOutput() const +{ + return mOutput; +} + +void AudioPolicyMix::clearOutput() +{ + mOutput.clear(); +} + +void AudioPolicyMix::setMix(AudioMix &mix) +{ + mMix = mix; +} + +android::AudioMix *AudioPolicyMix::getMix() +{ + return &mMix; +} + +status_t AudioPolicyMixCollection::registerMix(String8 address, AudioMix mix) +{ + ssize_t index = indexOfKey(address); + if (index >= 0) { + ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string()); + return BAD_VALUE; + } + sp<AudioPolicyMix> policyMix = new AudioPolicyMix(); + policyMix->setMix(mix); + add(address, policyMix); + return NO_ERROR; +} + +status_t AudioPolicyMixCollection::unregisterMix(String8 address) +{ + ssize_t index = indexOfKey(address); + if (index < 0) { + ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); + return BAD_VALUE; + } + + removeItemsAt(index); + return NO_ERROR; +} + +status_t AudioPolicyMixCollection::getAudioPolicyMix(String8 address, + sp<AudioPolicyMix> &policyMix) const +{ + ssize_t index = indexOfKey(address); + if (index < 0) { + ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); + return BAD_VALUE; + } + policyMix = valueAt(index); + return NO_ERROR; +} + +void AudioPolicyMixCollection::closeOutput(sp<SwAudioOutputDescriptor> &desc) +{ + for (size_t i = 0; i < size(); i++) { + sp<AudioPolicyMix> policyMix = valueAt(i); + if (policyMix->getOutput() == desc) { + policyMix->clearOutput(); + } + } +} + +status_t AudioPolicyMixCollection::getOutputForAttr(audio_attributes_t attributes, + sp<SwAudioOutputDescriptor> &desc) +{ + for (size_t i = 0; i < size(); i++) { + sp<AudioPolicyMix> policyMix = valueAt(i); + AudioMix *mix = policyMix->getMix(); + + if (mix->mMixType == MIX_TYPE_PLAYERS) { + for (size_t j = 0; j < mix->mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mUsage == attributes.usage) || + (RULE_EXCLUDE_ATTRIBUTE_USAGE == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mUsage != attributes.usage)) { + desc = policyMix->getOutput(); + break; + } + if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && + strncmp(attributes.tags + strlen("addr="), + mix->mRegistrationId.string(), + AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { + desc = policyMix->getOutput(); + break; + } + } + } else if (mix->mMixType == MIX_TYPE_RECORDERS) { + if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && + strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && + strncmp(attributes.tags + strlen("addr="), + mix->mRegistrationId.string(), + AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { + desc = policyMix->getOutput(); + } + } + if (desc != 0) { + desc->mPolicyMix = mix; + return NO_ERROR; + } + } + return BAD_VALUE; +} + +audio_devices_t AudioPolicyMixCollection::getDeviceAndMixForInputSource(audio_source_t inputSource, + audio_devices_t availDevices, + AudioMix **policyMix) +{ + for (size_t i = 0; i < size(); i++) { + AudioMix *mix = valueAt(i)->getMix(); + + if (mix->mMixType != MIX_TYPE_RECORDERS) { + continue; + } + for (size_t j = 0; j < mix->mCriteria.size(); j++) { + if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mSource == inputSource) || + (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mix->mCriteria[j].mRule && + mix->mCriteria[j].mAttr.mSource != inputSource)) { + if (availDevices & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + if (policyMix != NULL) { + *policyMix = mix; + } + return AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + } + } + } + return AUDIO_DEVICE_NONE; +} + +status_t AudioPolicyMixCollection::getInputMixForAttr(audio_attributes_t attr, AudioMix **policyMix) +{ + if (strncmp(attr.tags, "addr=", strlen("addr=")) != 0) { + return BAD_VALUE; + } + String8 address(attr.tags + strlen("addr=")); + + ssize_t index = indexOfKey(address); + if (index < 0) { + ALOGW("getInputForAttr() no policy for address %s", address.string()); + return BAD_VALUE; + } + sp<AudioPolicyMix> audioPolicyMix = valueAt(index); + AudioMix *mix = audioPolicyMix->getMix(); + + if (mix->mMixType != MIX_TYPE_PLAYERS) { + ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); + return BAD_VALUE; + } + *policyMix = mix; + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp new file mode 100644 index 0000000..e8191dd --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp @@ -0,0 +1,825 @@ +/* + * 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::AudioPort" +//#define LOG_NDEBUG 0 + +#include "AudioPort.h" +#include "HwModule.h" +#include "AudioGain.h" +#include "ConfigParsingUtils.h" +#include "audio_policy_conf.h" +#include <policy.h> + +namespace android { + +int32_t volatile AudioPort::mNextUniqueId = 1; + +// --- AudioPort class implementation + +AudioPort::AudioPort(const String8& name, audio_port_type_t type, + audio_port_role_t role) : + mName(name), mType(type), mRole(role), mFlags(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) +{ + mModule = module; +} + +audio_port_handle_t AudioPort::getNextUniqueId() +{ + return static_cast<audio_port_handle_t>(android_atomic_inc(&mNextUniqueId)); +} + +audio_module_handle_t AudioPort::getModuleHandle() const +{ + if (mModule == 0) { + return 0; + } + return mModule->mHandle; +} + +uint32_t AudioPort::getModuleVersion() const +{ + if (mModule == 0) { + return 0; + } + return mModule->mHalVersion; +} + +const char *AudioPort::getModuleName() const +{ + if (mModule == 0) { + return ""; + } + return mModule->mName; +} + +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.length() != 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); + for (size_t i = 0; i < mGains.size(); i++) { + mGains[i]->dump(fd, spaces + 2, i); + } + } +} + +void AudioPort::log(const char* indent) const +{ + ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole); +} + +// --- 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; + } +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp new file mode 100644 index 0000000..9ab1d61 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp @@ -0,0 +1,288 @@ +/* + * 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 "ConfigParsingUtils.h" +#include "AudioGain.h" +#include <hardware/audio.h> +#include <utils/Log.h> +#include <cutils/misc.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(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + devName); + } + devName = strtok(NULL, "|"); + } + return device; +} + +//static +void ConfigParsingUtils::loadHwModule(cnode *root, HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnable) +{ + status_t status = NAME_NOT_FOUND; + cnode *node; + sp<HwModule> module = new HwModule(root->name); + + node = config_find(root, DEVICES_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading device %s", node->name); + status_t tmpStatus = module->loadDevice(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, OUTPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading output %s", node->name); + status_t tmpStatus = module->loadOutput(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + node = config_find(root, INPUTS_TAG); + if (node != NULL) { + node = node->first_child; + while (node) { + ALOGV("loadHwModule() loading input %s", node->name); + status_t tmpStatus = module->loadInput(node); + if (status == NAME_NOT_FOUND || status == NO_ERROR) { + status = tmpStatus; + } + node = node->next; + } + } + loadGlobalConfig(root, module, availableInputDevices, availableOutputDevices, + defaultOutputDevices, isSpeakerDrcEnable); + + if (status == NO_ERROR) { + hwModules.add(module); + } +} + +//static +void ConfigParsingUtils::loadHwModules(cnode *root, HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled) +{ + cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); + if (node == NULL) { + return; + } + + node = node->first_child; + while (node) { + ALOGV("loadHwModules() loading module %s", node->name); + loadHwModule(node, hwModules, availableInputDevices, availableOutputDevices, + defaultOutputDevices, isSpeakerDrcEnabled); + node = node->next; + } +} + +//static +void ConfigParsingUtils::loadGlobalConfig(cnode *root, const sp<HwModule>& module, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevice, + bool &speakerDrcEnabled) +{ + cnode *node = config_find(root, GLOBAL_CONFIG_TAG); + + if (node == NULL) { + return; + } + DeviceVector declaredDevices; + if (module != NULL) { + declaredDevices = module->mDeclaredDevices; + } + + node = node->first_child; + while (node) { + if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { + availableOutputDevices.loadDevicesFromName((char *)node->value, + declaredDevices); + ALOGV("loadGlobalConfig() Attached Output Devices %08x", + availableOutputDevices.types()); + } else if (strcmp(DEFAULT_OUTPUT_DEVICE_TAG, node->name) == 0) { + audio_devices_t device = (audio_devices_t)stringToEnum( + sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + (char *)node->value); + if (device != AUDIO_DEVICE_NONE) { + defaultOutputDevice = new DeviceDescriptor(String8("default-output"), device); + } else { + ALOGW("loadGlobalConfig() default device not specified"); + } + ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", defaultOutputDevice->type()); + } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { + availableInputDevices.loadDevicesFromName((char *)node->value, + declaredDevices); + ALOGV("loadGlobalConfig() Available InputDevices %08x", availableInputDevices.types()); + } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { + speakerDrcEnabled = stringToBool((char *)node->value); + ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", speakerDrcEnabled); + } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) { + uint32_t major, minor; + sscanf((char *)node->value, "%u.%u", &major, &minor); + module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor); + ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u", + module->mHalVersion, major, minor); + } + node = node->next; + } +} + +//static +status_t ConfigParsingUtils::loadAudioPolicyConfig(const char *path, + HwModuleCollection &hwModules, + DeviceVector &availableInputDevices, + DeviceVector &availableOutputDevices, + sp<DeviceDescriptor> &defaultOutputDevices, + bool &isSpeakerDrcEnabled) +{ + cnode *root; + char *data; + + data = (char *)load_file(path, NULL); + if (data == NULL) { + return -ENODEV; + } + root = config_node("", ""); + config_load(root, data); + + loadHwModules(root, hwModules, + availableInputDevices, availableOutputDevices, + defaultOutputDevices, isSpeakerDrcEnabled); + // legacy audio_policy.conf files have one global_configuration section + loadGlobalConfig(root, hwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY), + availableInputDevices, availableOutputDevices, + defaultOutputDevices, isSpeakerDrcEnabled); + config_free(root); + free(root); + free(data); + + ALOGI("loadAudioPolicyConfig() loaded %s\n", path); + + return NO_ERROR; +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp new file mode 100644 index 0000000..9573583 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp @@ -0,0 +1,345 @@ +/* + * 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 "DeviceDescriptor.h" +#include "AudioGain.h" +#include "HwModule.h" +#include "ConfigParsingUtils.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), + mAddress(""), mDeviceType(type), mId(0) +{ + +} + +audio_port_handle_t DeviceDescriptor::getId() const +{ + return mId; +} + +void DeviceDescriptor::attach(const sp<HwModule>& module) +{ + AudioPort::attach(module); + mId = getNextUniqueId(); +} + +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)->type(); + } + 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->type()); + 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->type()); + } else { + ret = SortedVector::removeAt(ret); + if (ret >= 0) { + refreshTypes(); + } + } + return ret; +} + +audio_devices_t DeviceVector::getDevicesFromHwModule(audio_module_handle_t moduleHandle) const +{ + audio_devices_t devices = AUDIO_DEVICE_NONE; + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->getModuleHandle() == moduleHandle) { + devices |= itemAt(i)->type(); + } + } + return devices; +} + +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(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + devName); + if (type != AUDIO_DEVICE_NONE) { + devName = (char *)ConfigParsingUtils::enumToString(sDeviceNameToEnumTable, + ARRAY_SIZE(sDeviceNameToEnumTable), + type); + sp<DeviceDescriptor> dev = new DeviceDescriptor(String8(devName), 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)->type() == 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)->getId() == id) { + device = itemAt(i); + break; + } + } + return device; +} + +DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const +{ + DeviceVector devices; + bool isOutput = audio_is_output_devices(type); + type &= ~AUDIO_DEVICE_BIT_IN; + for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { + bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType); + audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN; + if ((isOutput == curIsOutput) && ((type & curType) != 0)) { + devices.add(itemAt(i)); + type &= ~curType; + ALOGV("DeviceVector::getDevicesFromType() for type %x found %p", + itemAt(i)->type(), 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)->type() == 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; +} + + +status_t DeviceVector::dump(int fd, const String8 &direction) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\n Available %s devices:\n", direction.string()); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + itemAt(i)->dump(fd, 2, i); + } + return NO_ERROR; +} + +audio_policy_dev_state_t DeviceVector::getDeviceConnectionState(const sp<DeviceDescriptor> &devDesc) const +{ + ssize_t index = indexOf(devDesc); + return index >= 0 ? AUDIO_POLICY_DEVICE_STATE_AVAILABLE : AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; +} + +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 != 0 ? mModule->mHandle : AUDIO_IO_HANDLE_NONE; + 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(sDeviceTypeToEnumTable, + ARRAY_SIZE(sDeviceTypeToEnumTable), + 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; +} + +void DeviceDescriptor::log() const +{ + ALOGI("Device id:%d type:0x%X:%s, addr:%s", + mId, + mDeviceType, + ConfigParsingUtils::enumToString( + sDeviceNameToEnumTable, ARRAY_SIZE(sDeviceNameToEnumTable), mDeviceType), + mAddress.string()); + + AudioPort::log(" "); +} + +}; // namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp new file mode 100644 index 0000000..33d838d --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/EffectDescriptor.cpp @@ -0,0 +1,192 @@ +/* + * 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::EffectDescriptor" +//#define LOG_NDEBUG 0 + +#include "EffectDescriptor.h" +#include <utils/String8.h> + +namespace android { + +status_t EffectDescriptor::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + String8 result; + + snprintf(buffer, SIZE, " I/O: %d\n", mIo); + result.append(buffer); + snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); + result.append(buffer); + snprintf(buffer, SIZE, " Session: %d\n", mSession); + result.append(buffer); + snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); + result.append(buffer); + snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); + result.append(buffer); + write(fd, result.string(), result.size()); + + return NO_ERROR; +} + +EffectDescriptorCollection::EffectDescriptorCollection() : + mTotalEffectsCpuLoad(0), + mTotalEffectsMemory(0) +{ + +} + +status_t EffectDescriptorCollection::registerEffect(const effect_descriptor_t *desc, + audio_io_handle_t io, + uint32_t strategy, + int session, + int id) +{ + if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { + ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", + desc->name, desc->memoryUsage); + return INVALID_OPERATION; + } + mTotalEffectsMemory += desc->memoryUsage; + ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", + desc->name, io, strategy, session, id); + ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); + + sp<EffectDescriptor> effectDesc = new EffectDescriptor(); + memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t)); + effectDesc->mIo = io; + effectDesc->mStrategy = static_cast<routing_strategy>(strategy); + effectDesc->mSession = session; + effectDesc->mEnabled = false; + + add(id, effectDesc); + + return NO_ERROR; +} + +status_t EffectDescriptorCollection::unregisterEffect(int id) +{ + ssize_t index = indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + sp<EffectDescriptor> effectDesc = valueAt(index); + + setEffectEnabled(effectDesc, false); + + if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) { + ALOGW("unregisterEffect() memory %d too big for total %d", + effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); + effectDesc->mDesc.memoryUsage = mTotalEffectsMemory; + } + mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage; + ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", + effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); + + removeItem(id); + + return NO_ERROR; +} + +status_t EffectDescriptorCollection::setEffectEnabled(int id, bool enabled) +{ + ssize_t index = indexOfKey(id); + if (index < 0) { + ALOGW("unregisterEffect() unknown effect ID %d", id); + return INVALID_OPERATION; + } + + return setEffectEnabled(valueAt(index), enabled); +} + + +status_t EffectDescriptorCollection::setEffectEnabled(const sp<EffectDescriptor> &effectDesc, + bool enabled) +{ + if (enabled == effectDesc->mEnabled) { + ALOGV("setEffectEnabled(%s) effect already %s", + enabled?"true":"false", enabled?"enabled":"disabled"); + return INVALID_OPERATION; + } + + if (enabled) { + if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { + ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", + effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10); + return INVALID_OPERATION; + } + mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); + } else { + if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) { + ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", + effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); + effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; + } + mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad; + ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); + } + effectDesc->mEnabled = enabled; + return NO_ERROR; +} + +bool EffectDescriptorCollection::isNonOffloadableEffectEnabled() +{ + for (size_t i = 0; i < size(); i++) { + sp<EffectDescriptor> effectDesc = valueAt(i); + if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) && + ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { + ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", + effectDesc->mDesc.name, effectDesc->mSession); + return true; + } + } + return false; +} + +uint32_t EffectDescriptorCollection::getMaxEffectsCpuLoad() const +{ + return MAX_EFFECTS_CPU_LOAD; +} + +uint32_t EffectDescriptorCollection::getMaxEffectsMemory() const +{ + return MAX_EFFECTS_MEMORY; +} + +status_t EffectDescriptorCollection::dump(int fd) +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", + (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); + write(fd, buffer, strlen(buffer)); + + snprintf(buffer, SIZE, "Registered effects:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + snprintf(buffer, SIZE, "- Effect %d dump:\n", keyAt(i)); + write(fd, buffer, strlen(buffer)); + valueAt(i)->dump(fd); + } + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp new file mode 100644 index 0000000..e955447 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp @@ -0,0 +1,373 @@ +/* + * 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 "HwModule.h" +#include "IOProfile.h" +#include "AudioGain.h" +#include "ConfigParsingUtils.h" +#include "audio_policy_conf.h" +#include <hardware/audio.h> +#include <policy.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); + + 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()); + + profile->attach(this); + 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); + + 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); + profile->attach(this); + 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, APM_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); + + node = root->first_child; + while (node) { + if (strcmp(node->name, APM_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); + + 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); + + profile->attach(this); + 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); + + 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); + + profile->attach(this); + 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); + } + } +} + +sp <HwModule> HwModuleCollection::getModuleFromName(const char *name) const +{ + sp <HwModule> module; + + for (size_t i = 0; i < size(); i++) + { + if (strcmp(itemAt(i)->mName, name) == 0) { + return itemAt(i); + } + } + return module; +} + + +sp <HwModule> HwModuleCollection::getModuleForDevice(audio_devices_t device) const +{ + sp <HwModule> module; + + for (size_t i = 0; i < size(); i++) { + if (itemAt(i)->mHandle == 0) { + continue; + } + if (audio_is_output_device(device)) { + for (size_t j = 0; j < itemAt(i)->mOutputProfiles.size(); j++) + { + if (itemAt(i)->mOutputProfiles[j]->mSupportedDevices.types() & device) { + return itemAt(i); + } + } + } else { + for (size_t j = 0; j < itemAt(i)->mInputProfiles.size(); j++) { + if (itemAt(i)->mInputProfiles[j]->mSupportedDevices.types() & + device & ~AUDIO_DEVICE_BIT_IN) { + return itemAt(i); + } + } + } + } + return module; +} + +sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device, + const char *device_address, + const char *device_name) const +{ + String8 address = (device_address == NULL) ? String8("") : String8(device_address); + // handle legacy remote submix case where the address was not always specified + if (device_distinguishes_on_address(device) && (address.length() == 0)) { + address = String8("0"); + } + + for (size_t i = 0; i < size(); i++) { + const sp<HwModule> hwModule = itemAt(i); + if (hwModule->mHandle == 0) { + continue; + } + DeviceVector deviceList = + hwModule->mDeclaredDevices.getDevicesFromTypeAddr(device, address); + if (!deviceList.isEmpty()) { + return deviceList.itemAt(0); + } + deviceList = hwModule->mDeclaredDevices.getDevicesFromType(device); + if (!deviceList.isEmpty()) { + return deviceList.itemAt(0); + } + } + + sp<DeviceDescriptor> devDesc = + new DeviceDescriptor(String8(device_name != NULL ? device_name : ""), device); + devDesc->mAddress = address; + return devDesc; +} + +status_t HwModuleCollection::dump(int fd) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\nHW Modules dump:\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); + write(fd, buffer, strlen(buffer)); + itemAt(i)->dump(fd); + } + return NO_ERROR; +} + +} //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp new file mode 100644 index 0000000..de6539c --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp @@ -0,0 +1,148 @@ +/* + * 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 "IOProfile.h" +#include "HwModule.h" +#include "AudioGain.h" + +namespace android { + +IOProfile::IOProfile(const String8& name, audio_port_role_t role) + : AudioPort(name, AUDIO_PORT_TYPE_MIX, role) +{ +} + +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) { + // just check types if multiple devices are selected + if (popcount(device & ~AUDIO_DEVICE_BIT_IN) > 1) { + if ((mSupportedDevices.types() & device) != device) { + return false; + } + } else if (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/common/managerdefinitions/src/SoundTriggerSession.cpp b/services/audiopolicy/common/managerdefinitions/src/SoundTriggerSession.cpp new file mode 100644 index 0000000..8ca3ae0 --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/SoundTriggerSession.cpp @@ -0,0 +1,45 @@ +/* + * 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::SoundTriggerSession" +//#define LOG_NDEBUG 0 + +#include "SoundTriggerSession.h" + + +namespace android { + +status_t SoundTriggerSessionCollection::acquireSession(audio_session_t session, + audio_io_handle_t ioHandle) +{ + add(session, ioHandle); + + return NO_ERROR; +} + +status_t SoundTriggerSessionCollection::releaseSession(audio_session_t session) +{ + ssize_t index = indexOfKey(session); + if (index < 0) { + ALOGW("acquireSoundTriggerSession() session %d not registered", session); + return BAD_VALUE; + } + + removeItem(session); + return NO_ERROR; +} + +}; //namespace android diff --git a/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp new file mode 100644 index 0000000..b682e2c --- /dev/null +++ b/services/audiopolicy/common/managerdefinitions/src/StreamDescriptor.cpp @@ -0,0 +1,162 @@ +/* + * 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::Volumes" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include "StreamDescriptor.h" +#include <utils/Log.h> +#include <utils/String8.h> + +namespace android { + +// --- 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) const +{ + device = Volume::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::clearCurrentVolumeIndex() +{ + mIndexCur.clear(); +} + +void StreamDescriptor::addCurrentVolumeIndex(audio_devices_t device, int index) +{ + mIndexCur.add(device, index); +} + +void StreamDescriptor::setVolumeIndexMin(int volIndexMin) +{ + mIndexMin = volIndexMin; +} + +void StreamDescriptor::setVolumeIndexMax(int volIndexMax) +{ + mIndexMax = volIndexMax; +} + +void StreamDescriptor::setVolumeCurvePoint(Volume::device_category deviceCategory, + const VolumeCurvePoint *point) +{ + mVolumeCurve[deviceCategory] = point; +} + +void StreamDescriptor::dump(int fd) const +{ + 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()); +} + +StreamDescriptorCollection::StreamDescriptorCollection() +{ + for (size_t stream = 0 ; stream < AUDIO_STREAM_CNT; stream++) { + add(static_cast<audio_stream_type_t>(stream), StreamDescriptor()); + } +} + +bool StreamDescriptorCollection::canBeMuted(audio_stream_type_t stream) +{ + return valueAt(stream).canBeMuted(); +} + +void StreamDescriptorCollection::clearCurrentVolumeIndex(audio_stream_type_t stream) +{ + editValueAt(stream).clearCurrentVolumeIndex(); +} + +void StreamDescriptorCollection::addCurrentVolumeIndex(audio_stream_type_t stream, + audio_devices_t device, int index) +{ + editValueAt(stream).addCurrentVolumeIndex(device, index); +} + +void StreamDescriptorCollection::setVolumeCurvePoint(audio_stream_type_t stream, + Volume::device_category deviceCategory, + const VolumeCurvePoint *point) +{ + editValueAt(stream).setVolumeCurvePoint(deviceCategory, point); +} + +const VolumeCurvePoint *StreamDescriptorCollection::getVolumeCurvePoint(audio_stream_type_t stream, + Volume::device_category deviceCategory) const +{ + return valueAt(stream).getVolumeCurvePoint(deviceCategory); +} + +void StreamDescriptorCollection::setVolumeIndexMin(audio_stream_type_t stream,int volIndexMin) +{ + return editValueAt(stream).setVolumeIndexMin(volIndexMin); +} + +void StreamDescriptorCollection::setVolumeIndexMax(audio_stream_type_t stream,int volIndexMax) +{ + return editValueAt(stream).setVolumeIndexMax(volIndexMax); +} + +status_t StreamDescriptorCollection::dump(int fd) const +{ + const size_t SIZE = 256; + char buffer[SIZE]; + + snprintf(buffer, SIZE, "\nStreams dump:\n"); + write(fd, buffer, strlen(buffer)); + snprintf(buffer, SIZE, + " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); + write(fd, buffer, strlen(buffer)); + for (size_t i = 0; i < size(); i++) { + snprintf(buffer, SIZE, " %02zu ", i); + write(fd, buffer, strlen(buffer)); + valueAt(i).dump(fd); + } + + return NO_ERROR; +} + +}; // namespace android diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h new file mode 100755 index 0000000..db0573f --- /dev/null +++ b/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h @@ -0,0 +1,171 @@ +/* + * 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. + */ + +#pragma once + +#include <AudioPolicyManagerObserver.h> +#include <RoutingStrategy.h> +#include <Volume.h> +#include <HwModule.h> +#include <DeviceDescriptor.h> +#include <system/audio.h> +#include <system/audio_policy.h> +#include <utils/Errors.h> +#include <utils/Vector.h> + +namespace android { + +/** + * This interface is dedicated to the policy manager that a Policy Engine shall implement. + */ +class AudioPolicyManagerInterface +{ +public: + /** + * Checks if the engine was correctly initialized. + * + * @return NO_ERROR if initialization has been done correctly, error code otherwise.. + */ + virtual status_t initCheck() = 0; + + /** + * Sets the Manager observer that allows the engine to retrieve information on collection + * of devices, streams, HwModules, ... + * + * @param[in] observer handle on the manager. + */ + virtual void setObserver(AudioPolicyManagerObserver *observer) = 0; + + /** + * Get the input device selected for a given input source. + * + * @param[in] inputSource to get the selected input device associated to + * + * @return selected input device for the given input source, may be none if error. + */ + virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const = 0; + + /** + * Get the output device associated to a given strategy. + * + * @param[in] stream type for which the selected ouput device is requested. + * + * @return selected ouput device for the given strategy, may be none if error. + */ + virtual audio_devices_t getDeviceForStrategy(routing_strategy stategy) const = 0; + + /** + * Get the strategy selected for a given stream type. + * + * @param[in] stream: for which the selected strategy followed by is requested. + * + * @return strategy to be followed. + */ + virtual routing_strategy getStrategyForStream(audio_stream_type_t stream) = 0; + + /** + * Get the strategy selected for a given usage. + * + * @param[in] usage to get the selected strategy followed by. + * + * @return strategy to be followed. + */ + virtual routing_strategy getStrategyForUsage(audio_usage_t usage) = 0; + + /** + * Set the Telephony Mode. + * + * @param[in] mode: Android Phone state (normal, ringtone, csv, in communication) + * + * @return NO_ERROR if Telephony Mode set correctly, error code otherwise. + */ + virtual status_t setPhoneState(audio_mode_t mode) = 0; + + /** + * Get the telephony Mode + * + * @return the current telephony mode + */ + virtual audio_mode_t getPhoneState() const = 0; + + /** + * Set Force Use config for a given usage. + * + * @param[in] usage for which a configuration shall be forced. + * @param[in] config wished to be forced for the given usage. + * + * @return NO_ERROR if the Force Use config was set correctly, error code otherwise (e.g. config not + * allowed a given usage...) + */ + virtual status_t setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) = 0; + + /** + * Get Force Use config for a given usage. + * + * @param[in] usage for which a configuration shall be forced. + * + * @return config wished to be forced for the given usage. + */ + virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const = 0; + + /** + * Set the connection state of device(s). + * + * @param[in] devDesc for which the state has changed. + * @param[in] state of availability of this(these) device(s). + * + * @return NO_ERROR if devices criterion updated correctly, error code otherwise. + */ + virtual status_t setDeviceConnectionState(const android::sp<android::DeviceDescriptor> devDesc, + audio_policy_dev_state_t state) = 0; + + /** + * Translate a volume index given by the UI to an amplification value in dB for a stream type + * and a device category. + * + * @param[in] deviceCategory for which the conversion is requested. + * @param[in] stream type for which the conversion is requested. + * @param[in] indexInUi index received from the UI to be translated. + * + * @return amplification value in dB matching the UI index for this given device and stream. + */ + virtual float volIndexToDb(Volume::device_category deviceCategory, audio_stream_type_t stream, + int indexInUi) = 0; + + /** + * Initialize the min / max index of volume applicable for a given stream type. These indexes + * will be used upon conversion of UI index to volume amplification. + * + * @param[in] stream type for which the indexes need to be set + * @param[in] indexMin Minimum index allowed for this stream. + * @param[in] indexMax Maximum index allowed for this stream. + */ + virtual status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) = 0; + + /** + * Initialize volume curves for each strategy and device category + * + * @param[in] isSpeakerDrcEnabled true on devices that use DRC on the DEVICE_CATEGORY_SPEAKER + path to boost soft sounds, used to adjust volume curves accordingly + */ + virtual void initializeVolumeCurves(bool isSpeakerDrcEnabled) = 0; + +protected: + virtual ~AudioPolicyManagerInterface() {} +}; + +}; // namespace android diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h new file mode 100755 index 0000000..6d43df2 --- /dev/null +++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h @@ -0,0 +1,62 @@ +/* + * 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. + */ + +#pragma once + +#include <AudioGain.h> +#include <AudioPort.h> +#include <AudioPatch.h> +#include <IOProfile.h> +#include <DeviceDescriptor.h> +#include <AudioInputDescriptor.h> +#include <AudioOutputDescriptor.h> +#include <AudioPolicyMix.h> +#include <SoundTriggerSession.h> +#include <StreamDescriptor.h> + +namespace android { + +/** + * This interface is an observer that the manager shall implement to allows e.g. the engine + * to access to policy pillars elements (like output / input descritors collections, + * HwModule collections, AudioMix, ... + */ +class AudioPolicyManagerObserver +{ +public: + virtual const AudioPatchCollection &getAudioPatches() const = 0; + + virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const = 0; + + virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const = 0; + + virtual const SwAudioOutputCollection &getOutputs() const = 0; + + virtual const AudioInputCollection &getInputs() const = 0; + + virtual const DeviceVector &getAvailableOutputDevices() const = 0; + + virtual const DeviceVector &getAvailableInputDevices() const = 0; + + virtual StreamDescriptorCollection &getStreamDescriptors() = 0; + + virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const = 0; + +protected: + virtual ~AudioPolicyManagerObserver() {} +}; + +}; // namespace android diff --git a/services/audiopolicy/enginedefault/Android.mk b/services/audiopolicy/enginedefault/Android.mk new file mode 100755 index 0000000..b0ae835 --- /dev/null +++ b/services/audiopolicy/enginedefault/Android.mk @@ -0,0 +1,48 @@ +LOCAL_PATH := $(call my-dir) + +# Component build +####################################################################### + +include $(CLEAR_VARS) + +LOCAL_SRC_FILES := \ + src/Engine.cpp \ + src/EngineInstance.cpp \ + src/Gains.cpp \ + + +audio_policy_engine_includes_common := \ + $(LOCAL_PATH)/include \ + $(TOPDIR)frameworks/av/services/audiopolicy/engine/interface + +LOCAL_CFLAGS += \ + -Wall \ + -Werror \ + -Wextra \ + +LOCAL_EXPORT_C_INCLUDE_DIRS := \ + $(audio_policy_engine_includes_common) + +LOCAL_C_INCLUDES := \ + $(audio_policy_engine_includes_common) \ + $(TARGET_OUT_HEADERS)/hw \ + $(call include-path-for, frameworks-av) \ + $(call include-path-for, audio-utils) \ + $(call include-path-for, bionic) \ + $(TOPDIR)frameworks/av/services/audiopolicy/common/include + + +LOCAL_MODULE := libaudiopolicyenginedefault +LOCAL_MODULE_TAGS := optional +LOCAL_STATIC_LIBRARIES := \ + libmedia_helper \ + libaudiopolicycomponents + +LOCAL_SHARED_LIBRARIES += \ + libcutils \ + libutils \ + libaudioutils \ + +include external/stlport/libstlport.mk + +include $(BUILD_SHARED_LIBRARY) diff --git a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h new file mode 100755 index 0000000..1e329f0 --- /dev/null +++ b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h @@ -0,0 +1,76 @@ +/* + * 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. + */ + +#pragma once + +class AudioPolicyManagerInterface; + +namespace android +{ +namespace audio_policy +{ + +class Engine; + +class EngineInstance +{ +protected: + EngineInstance(); + +public: + virtual ~EngineInstance(); + + /** + * Get Audio Policy Engine instance. + * + * @return pointer to Route Manager Instance object. + */ + static EngineInstance *getInstance(); + + /** + * Interface query. + * The first client of an interface of the policy engine will start the singleton. + * + * @tparam RequestedInterface: interface that the client is wishing to retrieve. + * + * @return interface handle. + */ + template <class RequestedInterface> + RequestedInterface *queryInterface() const; + +protected: + /** + * Get Audio Policy Engine instance. + * + * @return Audio Policy Engine singleton. + */ + Engine *getEngine() const; + +private: + /* Copy facilities are put private to disable copy. */ + EngineInstance(const EngineInstance &object); + EngineInstance &operator=(const EngineInstance &object); +}; + +/** + * Limit template instantation to supported type interfaces. + * Compile time error will claim if invalid interface is requested. + */ +template <> +AudioPolicyManagerInterface *EngineInstance::queryInterface() const; + +} // namespace audio_policy +} // namespace android diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp new file mode 100755 index 0000000..50f1609 --- /dev/null +++ b/services/audiopolicy/enginedefault/src/Engine.cpp @@ -0,0 +1,708 @@ +/* + * 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::AudioPolicyEngine" +//#define LOG_NDEBUG 0 + +//#define VERY_VERBOSE_LOGGING +#ifdef VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include "Engine.h" +#include "Gains.h" +#include <AudioPolicyManagerObserver.h> +#include <AudioPort.h> +#include <IOProfile.h> +#include <policy.h> +#include <utils/String8.h> +#include <utils/Log.h> + +namespace android +{ +namespace audio_policy +{ + +Engine::Engine() + : mManagerInterface(this), + mPhoneState(AUDIO_MODE_NORMAL), + mApmObserver(NULL) +{ + for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) { + mForceUse[i] = AUDIO_POLICY_FORCE_NONE; + } +} + +Engine::~Engine() +{ +} + +void Engine::setObserver(AudioPolicyManagerObserver *observer) +{ + ALOG_ASSERT(observer != NULL, "Invalid Audio Policy Manager observer"); + mApmObserver = observer; +} + +status_t Engine::initCheck() +{ + return (mApmObserver != NULL) ? NO_ERROR : NO_INIT; +} + +float Engine::volIndexToDb(Volume::device_category category, audio_stream_type_t streamType, + int indexInUi) +{ + const StreamDescriptor &streamDesc = mApmObserver->getStreamDescriptors().valueAt(streamType); + return Gains::volIndexToDb(category, streamDesc, indexInUi); +} + + +status_t Engine::initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) +{ + ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); + if (indexMin < 0 || indexMin >= indexMax) { + ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", + stream , indexMin, indexMax); + return BAD_VALUE; + } + mApmObserver->getStreamDescriptors().setVolumeIndexMin(stream, indexMin); + mApmObserver->getStreamDescriptors().setVolumeIndexMax(stream, indexMax); + return NO_ERROR; +} + +void Engine::initializeVolumeCurves(bool isSpeakerDrcEnabled) +{ + StreamDescriptorCollection &streams = mApmObserver->getStreamDescriptors(); + + for (int i = 0; i < AUDIO_STREAM_CNT; i++) { + for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) { + streams.setVolumeCurvePoint(static_cast<audio_stream_type_t>(i), + static_cast<Volume::device_category>(j), + Gains::sVolumeProfiles[i][j]); + } + } + + // Check availability of DRC on speaker path: if available, override some of the speaker curves + if (isSpeakerDrcEnabled) { + streams.setVolumeCurvePoint(AUDIO_STREAM_SYSTEM, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sDefaultSystemVolumeCurveDrc); + streams.setVolumeCurvePoint(AUDIO_STREAM_RING, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sSpeakerSonificationVolumeCurveDrc); + streams.setVolumeCurvePoint(AUDIO_STREAM_ALARM, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sSpeakerSonificationVolumeCurveDrc); + streams.setVolumeCurvePoint(AUDIO_STREAM_NOTIFICATION, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sSpeakerSonificationVolumeCurveDrc); + streams.setVolumeCurvePoint(AUDIO_STREAM_MUSIC, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sSpeakerMediaVolumeCurveDrc); + streams.setVolumeCurvePoint(AUDIO_STREAM_ACCESSIBILITY, Volume::DEVICE_CATEGORY_SPEAKER, + Gains::sSpeakerMediaVolumeCurveDrc); + } +} + + +status_t Engine::setPhoneState(audio_mode_t state) +{ + ALOGV("setPhoneState() state %d", state); + + if (state < 0 || state >= AUDIO_MODE_CNT) { + ALOGW("setPhoneState() invalid state %d", state); + return BAD_VALUE; + } + + if (state == mPhoneState ) { + ALOGW("setPhoneState() setting same state %d", state); + return BAD_VALUE; + } + + // store previous phone state for management of sonification strategy below + int oldState = mPhoneState; + mPhoneState = state; + StreamDescriptorCollection &streams = mApmObserver->getStreamDescriptors(); + // are we entering or starting a call + if (!is_state_in_call(oldState) && is_state_in_call(state)) { + ALOGV(" Entering call in setPhoneState()"); + for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) { + streams.setVolumeCurvePoint(AUDIO_STREAM_DTMF, static_cast<Volume::device_category>(j), + Gains::sVolumeProfiles[AUDIO_STREAM_VOICE_CALL][j]); + } + } else if (is_state_in_call(oldState) && !is_state_in_call(state)) { + ALOGV(" Exiting call in setPhoneState()"); + for (int j = 0; j < Volume::DEVICE_CATEGORY_CNT; j++) { + streams.setVolumeCurvePoint(AUDIO_STREAM_DTMF, static_cast<Volume::device_category>(j), + Gains::sVolumeProfiles[AUDIO_STREAM_DTMF][j]); + } + } + return NO_ERROR; +} + +status_t Engine::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) +{ + switch(usage) { + case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: + if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && + config != AUDIO_POLICY_FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); + return BAD_VALUE; + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_MEDIA: + if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && + config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_ANALOG_DOCK && + config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) { + ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); + return BAD_VALUE; + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_RECORD: + if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_NONE) { + ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); + return BAD_VALUE; + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_DOCK: + if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && + config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && + config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && + config != AUDIO_POLICY_FORCE_ANALOG_DOCK && + config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { + ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_SYSTEM: + if (config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { + ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); + } + mForceUse[usage] = config; + break; + case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: + if (config != AUDIO_POLICY_FORCE_NONE && + config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { + ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); + } + mForceUse[usage] = config; + break; + default: + ALOGW("setForceUse() invalid usage %d", usage); + break; + } + return NO_ERROR; +} + +routing_strategy Engine::getStrategyForStream(audio_stream_type_t stream) +{ + // stream to strategy mapping + switch (stream) { + case AUDIO_STREAM_VOICE_CALL: + case AUDIO_STREAM_BLUETOOTH_SCO: + return STRATEGY_PHONE; + case AUDIO_STREAM_RING: + case AUDIO_STREAM_ALARM: + return STRATEGY_SONIFICATION; + case AUDIO_STREAM_NOTIFICATION: + return STRATEGY_SONIFICATION_RESPECTFUL; + case AUDIO_STREAM_DTMF: + return STRATEGY_DTMF; + default: + ALOGE("unknown stream type %d", stream); + case AUDIO_STREAM_SYSTEM: + // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs + // while key clicks are played produces a poor result + case AUDIO_STREAM_MUSIC: + return STRATEGY_MEDIA; + case AUDIO_STREAM_ENFORCED_AUDIBLE: + return STRATEGY_ENFORCED_AUDIBLE; + case AUDIO_STREAM_TTS: + return STRATEGY_TRANSMITTED_THROUGH_SPEAKER; + case AUDIO_STREAM_ACCESSIBILITY: + return STRATEGY_ACCESSIBILITY; + case AUDIO_STREAM_REROUTING: + return STRATEGY_REROUTING; + } +} + +routing_strategy Engine::getStrategyForUsage(audio_usage_t usage) +{ + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); + + // usage to strategy mapping + switch (usage) { + case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: + if (outputs.isStreamActive(AUDIO_STREAM_RING) || + outputs.isStreamActive(AUDIO_STREAM_ALARM)) { + return STRATEGY_SONIFICATION; + } + if (isInCall()) { + return STRATEGY_PHONE; + } + return STRATEGY_ACCESSIBILITY; + + case AUDIO_USAGE_MEDIA: + case AUDIO_USAGE_GAME: + case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: + case AUDIO_USAGE_ASSISTANCE_SONIFICATION: + return STRATEGY_MEDIA; + + case AUDIO_USAGE_VOICE_COMMUNICATION: + return STRATEGY_PHONE; + + case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: + return STRATEGY_DTMF; + + case AUDIO_USAGE_ALARM: + case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: + return STRATEGY_SONIFICATION; + + case AUDIO_USAGE_NOTIFICATION: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: + case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: + case AUDIO_USAGE_NOTIFICATION_EVENT: + return STRATEGY_SONIFICATION_RESPECTFUL; + + case AUDIO_USAGE_UNKNOWN: + default: + return STRATEGY_MEDIA; + } +} + +audio_devices_t Engine::getDeviceForStrategy(routing_strategy strategy) const +{ + const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices(); + const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices(); + + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); + + uint32_t device = AUDIO_DEVICE_NONE; + uint32_t availableOutputDevicesType = availableOutputDevices.types(); + + switch (strategy) { + + case STRATEGY_TRANSMITTED_THROUGH_SPEAKER: + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER; + if (!device) { + ALOGE("getDeviceForStrategy() no device found for "\ + "STRATEGY_TRANSMITTED_THROUGH_SPEAKER"); + } + break; + + case STRATEGY_SONIFICATION_RESPECTFUL: + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_SONIFICATION); + } else if (outputs.isStreamActiveRemotely(AUDIO_STREAM_MUSIC, + SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing on a remote device, use the the sonification behavior. + // Note that we test this usecase before testing if media is playing because + // the isStreamActive() method only informs about the activity of a stream, not + // if it's for local playback. Note also that we use the same delay between both tests + device = getDeviceForStrategy(STRATEGY_SONIFICATION); + //user "safe" speaker if available instead of normal speaker to avoid triggering + //other acoustic safety mechanisms for notification + if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) + device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; + } else if (outputs.isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { + // while media is playing (or has recently played), use the same device + device = getDeviceForStrategy(STRATEGY_MEDIA); + } else { + // when media is not playing anymore, fall back on the sonification behavior + device = getDeviceForStrategy(STRATEGY_SONIFICATION); + //user "safe" speaker if available instead of normal speaker to avoid triggering + //other acoustic safety mechanisms for notification + if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) + device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; + } + break; + + case STRATEGY_DTMF: + if (!isInCall()) { + // when off call, DTMF strategy follows the same rules as MEDIA strategy + device = getDeviceForStrategy(STRATEGY_MEDIA); + break; + } + // when in call, DTMF and PHONE strategies follow the same rules + // FALL THROUGH + + case STRATEGY_PHONE: + // Force use of only devices on primary output if: + // - in call AND + // - cannot route from voice call RX OR + // - audio HAL version is < 3.0 and TX device is on the primary HW module + if (getPhoneState() == AUDIO_MODE_IN_CALL) { + audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); + sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput(); + audio_devices_t availPrimaryInputDevices = + availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle()); + audio_devices_t availPrimaryOutputDevices = + primaryOutput->supportedDevices() & availableOutputDevices.types(); + + if (((availableInputDevices.types() & + AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || + (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) && + (primaryOutput->getAudioPort()->getModuleVersion() < + AUDIO_DEVICE_API_VERSION_3_0))) { + availableOutputDevicesType = availPrimaryOutputDevices; + } + } + // for phone strategy, we first consider the forced use and then the available devices by order + // of priority + switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { + case AUDIO_POLICY_FORCE_BT_SCO: + if (!isInCall() || strategy != STRATEGY_DTMF) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; + if (device) break; + } + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; + if (device) break; + // if SCO device is requested but no SCO device is available, fall back to default case + // FALL THROUGH + + default: // FORCE_NONE + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP + if (!isInCall() && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (outputs.getA2dpOutput() != 0)) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + if (device) break; + } + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + if (!isInCall()) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_EARPIECE; + if (device) break; + device = mApmObserver->getDefaultOutputDevice()->type(); + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); + } + break; + + case AUDIO_POLICY_FORCE_SPEAKER: + // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to + // A2DP speaker when forcing to speaker output + if (!isInCall() && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (outputs.getA2dpOutput() != 0)) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + if (device) break; + } + if (!isInCall()) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + if (device) break; + } + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE; + if (device) break; + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER; + if (device) break; + device = mApmObserver->getDefaultOutputDevice()->type(); + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); + } + break; + } + break; + + case STRATEGY_SONIFICATION: + + // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by + // handleIncallSonification(). + if (isInCall()) { + device = getDeviceForStrategy(STRATEGY_PHONE); + break; + } + // FALL THROUGH + + case STRATEGY_ENFORCED_AUDIBLE: + // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION + // except: + // - when in call where it doesn't default to STRATEGY_PHONE behavior + // - in countries where not enforced in which case it follows STRATEGY_MEDIA + + if ((strategy == STRATEGY_SONIFICATION) || + (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { + device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER; + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); + } + } + // The second device used for sonification is the same as the device used by media strategy + // FALL THROUGH + + // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now + case STRATEGY_ACCESSIBILITY: + if (strategy == STRATEGY_ACCESSIBILITY) { + // do not route accessibility prompts to a digital output currently configured with a + // compressed format as they would likely not be mixed and dropped. + for (size_t i = 0; i < outputs.size(); i++) { + sp<AudioOutputDescriptor> desc = outputs.valueAt(i); + audio_devices_t devices = desc->device() & + (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC); + if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) && + devices != AUDIO_DEVICE_NONE) { + availableOutputDevicesType = availableOutputDevices.types() & ~devices; + } + } + } + // FALL THROUGH + + case STRATEGY_REROUTING: + case STRATEGY_MEDIA: { + uint32_t device2 = AUDIO_DEVICE_NONE; + if (strategy != STRATEGY_SONIFICATION) { + // no sonification on remote submix (e.g. WFD) + if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) { + device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && + (outputs.getA2dpOutput() != 0)) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; + } + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; + } + if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { + // no sonification on aux digital (e.g. HDMI) + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL; + } + if ((device2 == AUDIO_DEVICE_NONE) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; + } + if (device2 == AUDIO_DEVICE_NONE) { + device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER; + } + int device3 = AUDIO_DEVICE_NONE; + if (strategy == STRATEGY_MEDIA) { + // ARC, SPDIF and AUX_LINE can co-exist with others. + device3 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HDMI_ARC; + device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPDIF); + device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_LINE); + } + + device2 |= device3; + // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or + // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise + device |= device2; + + // If hdmi system audio mode is on, remove speaker out of output list. + if ((strategy == STRATEGY_MEDIA) && + (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == + AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { + device &= ~AUDIO_DEVICE_OUT_SPEAKER; + } + + if (device) break; + device = mApmObserver->getDefaultOutputDevice()->type(); + if (device == AUDIO_DEVICE_NONE) { + ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); + } + } break; + + default: + ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); + break; + } + + ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); + return device; +} + + +audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const +{ + const DeviceVector &availableOutputDevices = mApmObserver->getAvailableOutputDevices(); + const DeviceVector &availableInputDevices = mApmObserver->getAvailableInputDevices(); + const SwAudioOutputCollection &outputs = mApmObserver->getOutputs(); + audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; + + uint32_t device = AUDIO_DEVICE_NONE; + + switch (inputSource) { + case AUDIO_SOURCE_VOICE_UPLINK: + if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + break; + } + break; + + case AUDIO_SOURCE_DEFAULT: + case AUDIO_SOURCE_MIC: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { + device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; + } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) && + (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + + case AUDIO_SOURCE_VOICE_COMMUNICATION: + // Allow only use of devices on primary input if in call and HAL does not support routing + // to voice call path. + if ((getPhoneState() == AUDIO_MODE_IN_CALL) && + (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { + sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput(); + availableDeviceTypes = + availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle()) + & ~AUDIO_DEVICE_BIT_IN; + } + + switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { + case AUDIO_POLICY_FORCE_BT_SCO: + // if SCO device is requested but no SCO device is available, fall back to default case + if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + break; + } + // FALL THROUGH + + default: // FORCE_NONE + if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + + case AUDIO_POLICY_FORCE_SPEAKER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + } + break; + + case AUDIO_SOURCE_VOICE_RECOGNITION: + case AUDIO_SOURCE_HOTWORD: + if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && + availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { + device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { + device = AUDIO_DEVICE_IN_WIRED_HEADSET; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { + device = AUDIO_DEVICE_IN_USB_DEVICE; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_CAMCORDER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { + device = AUDIO_DEVICE_IN_BACK_MIC; + } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { + device = AUDIO_DEVICE_IN_BUILTIN_MIC; + } + break; + case AUDIO_SOURCE_VOICE_DOWNLINK: + case AUDIO_SOURCE_VOICE_CALL: + if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { + device = AUDIO_DEVICE_IN_VOICE_CALL; + } + break; + case AUDIO_SOURCE_REMOTE_SUBMIX: + if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + } + break; + case AUDIO_SOURCE_FM_TUNER: + if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) { + device = AUDIO_DEVICE_IN_FM_TUNER; + } + break; + default: + ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); + break; + } + ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); + return device; +} + +template <> +AudioPolicyManagerInterface *Engine::queryInterface() +{ + return &mManagerInterface; +} + +} // namespace audio_policy +} // namespace android + + diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h new file mode 100755 index 0000000..56a4748 --- /dev/null +++ b/services/audiopolicy/enginedefault/src/Engine.h @@ -0,0 +1,158 @@ +/* + * 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. + */ + +#pragma once + + +#include "AudioPolicyManagerInterface.h" +#include "Gains.h" +#include <AudioGain.h> +#include <policy.h> + +namespace android +{ + +class AudioPolicyManagerObserver; + +namespace audio_policy +{ + +class Engine +{ +public: + Engine(); + virtual ~Engine(); + + template <class RequestedInterface> + RequestedInterface *queryInterface(); + +private: + /// Interface members + class ManagerInterfaceImpl : public AudioPolicyManagerInterface + { + public: + ManagerInterfaceImpl(Engine *policyEngine) + : mPolicyEngine(policyEngine) {} + + virtual void setObserver(AudioPolicyManagerObserver *observer) + { + mPolicyEngine->setObserver(observer); + } + virtual status_t initCheck() + { + return mPolicyEngine->initCheck(); + } + virtual audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const + { + return mPolicyEngine->getDeviceForInputSource(inputSource); + } + virtual audio_devices_t getDeviceForStrategy(routing_strategy strategy) const + { + return mPolicyEngine->getDeviceForStrategy(strategy); + } + virtual routing_strategy getStrategyForStream(audio_stream_type_t stream) + { + return mPolicyEngine->getStrategyForStream(stream); + } + virtual routing_strategy getStrategyForUsage(audio_usage_t usage) + { + return mPolicyEngine->getStrategyForUsage(usage); + } + virtual status_t setPhoneState(audio_mode_t mode) + { + return mPolicyEngine->setPhoneState(mode); + } + virtual audio_mode_t getPhoneState() const + { + return mPolicyEngine->getPhoneState(); + } + virtual status_t setForceUse(audio_policy_force_use_t usage, + audio_policy_forced_cfg_t config) + { + return mPolicyEngine->setForceUse(usage, config); + } + virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const + { + return mPolicyEngine->getForceUse(usage); + } + virtual status_t setDeviceConnectionState(const sp<DeviceDescriptor> /*devDesc*/, + audio_policy_dev_state_t /*state*/) + { + return NO_ERROR; + } + virtual status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax) + { + return mPolicyEngine->initStreamVolume(stream, indexMin, indexMax); + } + virtual void initializeVolumeCurves(bool isSpeakerDrcEnabled) + { + return mPolicyEngine->initializeVolumeCurves(isSpeakerDrcEnabled); + } + virtual float volIndexToDb(Volume::device_category deviceCategory, + audio_stream_type_t stream,int indexInUi) + { + return mPolicyEngine->volIndexToDb(deviceCategory, stream, indexInUi); + } + private: + Engine *mPolicyEngine; + } mManagerInterface; + +private: + /* Copy facilities are put private to disable copy. */ + Engine(const Engine &object); + Engine &operator=(const Engine &object); + + void setObserver(AudioPolicyManagerObserver *observer); + + status_t initCheck(); + + inline bool isInCall() const + { + return is_state_in_call(mPhoneState); + } + + status_t setPhoneState(audio_mode_t mode); + audio_mode_t getPhoneState() const + { + return mPhoneState; + } + status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); + audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) const + { + return mForceUse[usage]; + } + status_t setDefaultDevice(audio_devices_t device); + + routing_strategy getStrategyForStream(audio_stream_type_t stream); + routing_strategy getStrategyForUsage(audio_usage_t usage); + audio_devices_t getDeviceForStrategy(routing_strategy strategy) const; + audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const; + + float volIndexToDb(Volume::device_category category, + audio_stream_type_t stream, int indexInUi); + status_t initStreamVolume(audio_stream_type_t stream, int indexMin, int indexMax); + void initializeVolumeCurves(bool isSpeakerDrcEnabled); + + audio_mode_t mPhoneState; /**< current phone state. */ + + /** current forced use configuration. */ + audio_policy_forced_cfg_t mForceUse[AUDIO_POLICY_FORCE_USE_CNT]; + + AudioPolicyManagerObserver *mApmObserver; +}; +} // namespace audio_policy +} // namespace android + diff --git a/services/audiopolicy/enginedefault/src/EngineInstance.cpp b/services/audiopolicy/enginedefault/src/EngineInstance.cpp new file mode 100755 index 0000000..17e9832 --- /dev/null +++ b/services/audiopolicy/enginedefault/src/EngineInstance.cpp @@ -0,0 +1,54 @@ +/* + * 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 <AudioPolicyManagerInterface.h> +#include "AudioPolicyEngineInstance.h" +#include "Engine.h" + +namespace android +{ +namespace audio_policy +{ + +EngineInstance::EngineInstance() +{ +} + +EngineInstance *EngineInstance::getInstance() +{ + static EngineInstance instance; + return &instance; +} + +EngineInstance::~EngineInstance() +{ +} + +Engine *EngineInstance::getEngine() const +{ + static Engine engine; + return &engine; +} + +template <> +AudioPolicyManagerInterface *EngineInstance::queryInterface() const +{ + return getEngine()->queryInterface<AudioPolicyManagerInterface>(); +} + +} // namespace audio_policy +} // namespace android + diff --git a/services/audiopolicy/enginedefault/src/Gains.cpp b/services/audiopolicy/enginedefault/src/Gains.cpp new file mode 100644 index 0000000..78f2909 --- /dev/null +++ b/services/audiopolicy/enginedefault/src/Gains.cpp @@ -0,0 +1,255 @@ +/* + * 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 "Gains.h" +#include <Volume.h> +#include <math.h> +#include <utils/String8.h> + +namespace android { + +// Enginedefault +const VolumeCurvePoint +Gains::sDefaultVolumeCurve[Volume::VOLCNT] = { + {1, -49.5f}, {33, -33.5f}, {66, -17.0f}, {100, 0.0f} +}; + + +const VolumeCurvePoint +Gains::sDefaultMediaVolumeCurve[Volume::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -17.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sExtMediaSystemVolumeCurve[Volume::VOLCNT] = { + {1, -58.0f}, {20, -40.0f}, {60, -21.0f}, {100, -10.0f} +}; + +const VolumeCurvePoint +Gains::sSpeakerMediaVolumeCurve[Volume::VOLCNT] = { + {1, -56.0f}, {20, -34.0f}, {60, -11.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sSpeakerMediaVolumeCurveDrc[Volume::VOLCNT] = { + {1, -55.0f}, {20, -43.0f}, {86, -12.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sSpeakerSonificationVolumeCurve[Volume::VOLCNT] = { + {1, -29.7f}, {33, -20.1f}, {66, -10.2f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sSpeakerSonificationVolumeCurveDrc[Volume::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 +Gains::sDefaultSystemVolumeCurve[Volume::VOLCNT] = { + {1, -24.0f}, {33, -18.0f}, {66, -12.0f}, {100, -6.0f} +}; + +const VolumeCurvePoint +Gains::sDefaultSystemVolumeCurveDrc[Volume::VOLCNT] = { + {1, -34.0f}, {33, -24.0f}, {66, -15.0f}, {100, -6.0f} +}; + +const VolumeCurvePoint +Gains::sHeadsetSystemVolumeCurve[Volume::VOLCNT] = { + {1, -30.0f}, {33, -26.0f}, {66, -22.0f}, {100, -18.0f} +}; + +const VolumeCurvePoint +Gains::sDefaultVoiceVolumeCurve[Volume::VOLCNT] = { + {0, -42.0f}, {33, -28.0f}, {66, -14.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sSpeakerVoiceVolumeCurve[Volume::VOLCNT] = { + {0, -24.0f}, {33, -16.0f}, {66, -8.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sLinearVolumeCurve[Volume::VOLCNT] = { + {0, -96.0f}, {33, -68.0f}, {66, -34.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint +Gains::sSilentVolumeCurve[Volume::VOLCNT] = { + {0, -96.0f}, {1, -96.0f}, {2, -96.0f}, {100, -96.0f} +}; + +const VolumeCurvePoint +Gains::sFullScaleVolumeCurve[Volume::VOLCNT] = { + {0, 0.0f}, {1, 0.0f}, {2, 0.0f}, {100, 0.0f} +}; + +const VolumeCurvePoint *Gains::sVolumeProfiles[AUDIO_STREAM_CNT] + [Volume::DEVICE_CATEGORY_CNT] = { + { // AUDIO_STREAM_VOICE_CALL + Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_SYSTEM + Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_RING + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_MUSIC + Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ALARM + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_NOTIFICATION + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerSonificationVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_BLUETOOTH_SCO + Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerVoiceVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultVoiceVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ENFORCED_AUDIBLE + Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_DTMF + Gains::sHeadsetSystemVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultSystemVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sExtMediaSystemVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_TTS + // "Transmitted Through Speaker": always silent except on DEVICE_CATEGORY_SPEAKER + Gains::sSilentVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sLinearVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sSilentVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sSilentVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_ACCESSIBILITY + Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sSpeakerMediaVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sDefaultMediaVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sDefaultMediaVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_REROUTING + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, + { // AUDIO_STREAM_PATCH + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_HEADSET + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_SPEAKER + Gains::sFullScaleVolumeCurve, // DEVICE_CATEGORY_EARPIECE + Gains::sFullScaleVolumeCurve // DEVICE_CATEGORY_EXT_MEDIA + }, +}; + +//static +float Gains::volIndexToDb(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi) +{ + const VolumeCurvePoint *curve = streamDesc.getVolumeCurvePoint(deviceCategory); + + // the volume index in the UI is relative to the min and max volume indices for this stream type + int nbSteps = 1 + curve[Volume::VOLMAX].mIndex - + curve[Volume::VOLMIN].mIndex; + int volIdx = (nbSteps * (indexInUi - streamDesc.getVolumeIndexMin())) / + (streamDesc.getVolumeIndexMax() - streamDesc.getVolumeIndexMin()); + + // find what part of the curve this index volume belongs to, or if it's out of bounds + int segment = 0; + if (volIdx < curve[Volume::VOLMIN].mIndex) { // out of bounds + return VOLUME_MIN_DB; + } else if (volIdx < curve[Volume::VOLKNEE1].mIndex) { + segment = 0; + } else if (volIdx < curve[Volume::VOLKNEE2].mIndex) { + segment = 1; + } else if (volIdx <= curve[Volume::VOLMAX].mIndex) { + segment = 2; + } else { // out of bounds + return 0.0f; + } + + // 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)) ); + + ALOGVV("VOLUME vol index=[%d %d %d], dB=[%.1f %.1f %.1f]", + curve[segment].mIndex, volIdx, + curve[segment+1].mIndex, + curve[segment].mDBAttenuation, + decibels, + curve[segment+1].mDBAttenuation); + + return decibels; +} + + +//static +float Gains::volIndexToAmpl(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi) +{ + return Volume::DbToAmpl(volIndexToDb(deviceCategory, streamDesc, indexInUi)); +} + + + +}; // namespace android diff --git a/services/audiopolicy/enginedefault/src/Gains.h b/services/audiopolicy/enginedefault/src/Gains.h new file mode 100644 index 0000000..7620b7d --- /dev/null +++ b/services/audiopolicy/enginedefault/src/Gains.h @@ -0,0 +1,64 @@ +/* + * 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. + */ + +#pragma once + +#include <StreamDescriptor.h> +#include <utils/KeyedVector.h> +#include <system/audio.h> +#include <utils/Errors.h> +#include <utils/RefBase.h> + +namespace android { + +class StreamDescriptor; + +class Gains +{ +public : + static float volIndexToAmpl(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi); + + static float volIndexToDb(Volume::device_category deviceCategory, + const StreamDescriptor& streamDesc, + int indexInUi); + + // default volume curve + static const VolumeCurvePoint sDefaultVolumeCurve[Volume::VOLCNT]; + // default volume curve for media strategy + static const VolumeCurvePoint sDefaultMediaVolumeCurve[Volume::VOLCNT]; + // volume curve for non-media audio on ext media outputs (HDMI, Line, etc) + static const VolumeCurvePoint sExtMediaSystemVolumeCurve[Volume::VOLCNT]; + // volume curve for media strategy on speakers + static const VolumeCurvePoint sSpeakerMediaVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sSpeakerMediaVolumeCurveDrc[Volume::VOLCNT]; + // volume curve for sonification strategy on speakers + static const VolumeCurvePoint sSpeakerSonificationVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sSpeakerSonificationVolumeCurveDrc[Volume::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sDefaultSystemVolumeCurveDrc[Volume::VOLCNT]; + static const VolumeCurvePoint sHeadsetSystemVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sDefaultVoiceVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sSpeakerVoiceVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sLinearVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sSilentVolumeCurve[Volume::VOLCNT]; + static const VolumeCurvePoint sFullScaleVolumeCurve[Volume::VOLCNT]; + // default volume curves per stream and device category. See initializeVolumeCurves() + static const VolumeCurvePoint *sVolumeProfiles[AUDIO_STREAM_CNT][Volume::DEVICE_CATEGORY_CNT]; +}; + +}; // namespace android diff --git a/services/audiopolicy/AudioPolicyFactory.cpp b/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/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp index 6ebd0ed..35e80f7 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 @@ -24,20 +24,11 @@ #define ALOGVV(a...) do { } while(0) #endif -// A device mask for all audio input devices that are considered "virtual" when evaluating -// active inputs in getActiveInput() -#define APM_AUDIO_IN_DEVICE_VIRTUAL_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX|AUDIO_DEVICE_IN_FM_TUNER) -// A device mask for all audio output devices that are considered "remote" when evaluating -// active output devices in isStreamActiveRemotely() -#define APM_AUDIO_OUT_DEVICE_REMOTE_ALL AUDIO_DEVICE_OUT_REMOTE_SUBMIX -// A device mask for all audio input and output devices where matching inputs/outputs on device -// type alone is not enough: the address must match too -#define APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL (AUDIO_DEVICE_IN_REMOTE_SUBMIX | \ - AUDIO_DEVICE_OUT_REMOTE_SUBMIX) - #include <inttypes.h> #include <math.h> +#include <AudioPolicyManagerInterface.h> +#include <AudioPolicyEngineInstance.h> #include <cutils/properties.h> #include <utils/Log.h> #include <hardware/audio.h> @@ -47,188 +38,36 @@ #include <soundtrigger/SoundTrigger.h> #include "AudioPolicyManager.h" #include "audio_policy_conf.h" +#include <ConfigParsingUtils.h> +#include <policy.h> 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 // ---------------------------------------------------------------------------- status_t AudioPolicyManager::setDeviceConnectionState(audio_devices_t device, - audio_policy_dev_state_t state, - const char *device_address) + audio_policy_dev_state_t state, + const char *device_address, + const char *device_name) { - return setDeviceConnectionStateInt(device, state, device_address); + return setDeviceConnectionStateInt(device, state, device_address, device_name); } status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address) + const char *device_address, + const char *device_name) { - ALOGV("setDeviceConnectionState() device: %x, state %d, address %s", - device, state, device_address != NULL ? device_address : ""); + ALOGV("setDeviceConnectionStateInt() device: 0x%X, state %d, address %s name %s", +- device, state, device_address, device_name); // connect/disconnect only 1 device at a time if (!audio_is_output_device(device) && !audio_is_input_device(device)) return BAD_VALUE; - sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address); + sp<DeviceDescriptor> devDesc = + mHwModules.getDeviceDescriptor(device, device_address, device_name); // handle output devices if (audio_is_output_device(device)) { @@ -252,15 +91,14 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, // register new device as available index = mAvailableOutputDevices.add(devDesc); if (index >= 0) { - sp<HwModule> module = getModuleForDevice(device); + sp<HwModule> module = mHwModules.getModuleForDevice(device); if (module == 0) { ALOGD("setDeviceConnectionState() could not find HW module for device %08x", device); mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } - mAvailableOutputDevices[index]->mId = nextUniqueId(); - mAvailableOutputDevices[index]->mModule = module; + mAvailableOutputDevices[index]->attach(module); } else { return NO_MEMORY; } @@ -269,14 +107,16 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, mAvailableOutputDevices.remove(devDesc); return INVALID_OPERATION; } + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); + // outputs should never be empty here ALOG_ASSERT(outputs.size() != 0, "setDeviceConnectionState():" "checkOutputsForDevice() returned no outputs but status OK"); 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()); @@ -291,7 +131,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, ALOGV("setDeviceConnectionState() disconnecting output device %x", device); - // Set Disconnect to HALs + // Send Disconnect to HALs AudioParameter param = AudioParameter(devDesc->mAddress); param.addInt(String8(AUDIO_PARAMETER_DEVICE_DISCONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); @@ -300,6 +140,9 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, mAvailableOutputDevices.remove(devDesc); checkOutputsForDevice(devDesc, state, outputs, devDesc->mAddress); + + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); } break; default: @@ -314,7 +157,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, // outputs must be closed after checkOutputForAllStrategies() is executed if (!outputs.isEmpty()) { for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); // close unused outputs after device disconnection or direct outputs that have been // opened by checkOutputsForDevice() to query dynamic parameters if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) || @@ -328,23 +171,22 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, } updateDevicesAndOutputs(); - if (mPhoneState == AUDIO_MODE_IN_CALL) { + if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - audio_devices_t newDevice = getNewOutputDevice(mOutputs.keyAt(i), - true /*fromCache*/); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); + if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) { + audio_devices_t newDevice = getNewOutputDevice(desc, true /*fromCache*/); // do not force device change on duplicated output because if device is 0, it will // also force a device 0 for the two outputs it is duplicated to which may override // a valid device selection on those outputs. - bool force = !mOutputs.valueAt(i)->isDuplicated() - && (!deviceDistinguishesOnAddress(device) + bool force = !desc->isDuplicated() + && (!device_distinguishes_on_address(device) // always force when disconnecting (a non-duplicated device) || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)); - setOutputDevice(output, newDevice, force, 0); + setOutputDevice(desc, newDevice, force, 0); } } @@ -365,7 +207,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, ALOGW("setDeviceConnectionState() device already connected: %d", device); return INVALID_OPERATION; } - sp<HwModule> module = getModuleForDevice(device); + sp<HwModule> module = mHwModules.getModuleForDevice(device); if (module == NULL) { ALOGW("setDeviceConnectionState(): could not find HW module for device %08x", device); @@ -377,8 +219,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, index = mAvailableInputDevices.add(devDesc); if (index >= 0) { - mAvailableInputDevices[index]->mId = nextUniqueId(); - mAvailableInputDevices[index]->mModule = module; + mAvailableInputDevices[index]->attach(module); } else { return NO_MEMORY; } @@ -388,6 +229,8 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, param.addInt(String8(AUDIO_PARAMETER_DEVICE_CONNECT), device); mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString()); + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); } break; // handle input device disconnection @@ -407,6 +250,8 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, checkInputsForDevice(device, state, inputs, devDesc->mAddress); mAvailableInputDevices.remove(devDesc); + // Propagate device availability to Engine + mEngine->setDeviceConnectionState(devDesc, state); } break; default: @@ -416,7 +261,7 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, closeAllInputs(); - if (mPhoneState == AUDIO_MODE_IN_CALL) { + if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/); updateCallRouting(newDevice); } @@ -430,9 +275,10 @@ status_t AudioPolicyManager::setDeviceConnectionStateInt(audio_devices_t device, } audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devices_t device, - const char *device_address) + const char *device_address) { - sp<DeviceDescriptor> devDesc = getDeviceDescriptor(device, device_address); + sp<DeviceDescriptor> devDesc = mHwModules.getDeviceDescriptor(device, device_address, ""); + DeviceVector *deviceVector; if (audio_is_output_device(device)) { @@ -443,43 +289,7 @@ audio_policy_dev_state_t AudioPolicyManager::getDeviceConnectionState(audio_devi ALOGW("getDeviceConnectionState() invalid device type %08x", device); return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; } - - ssize_t index = deviceVector->indexOf(devDesc); - if (index >= 0) { - return AUDIO_POLICY_DEVICE_STATE_AVAILABLE; - } else { - return AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE; - } -} - -sp<AudioPolicyManager::DeviceDescriptor> AudioPolicyManager::getDeviceDescriptor( - const audio_devices_t device, - const char *device_address) -{ - String8 address = (device_address == NULL) ? String8("") : String8(device_address); - // handle legacy remote submix case where the address was not always specified - if (deviceDistinguishesOnAddress(device) && (address.length() == 0)) { - address = String8("0"); - } - - for (size_t i = 0; i < mHwModules.size(); i++) { - if (mHwModules[i]->mHandle == 0) { - continue; - } - DeviceVector deviceList = - mHwModules[i]->mDeclaredDevices.getDevicesFromTypeAddr(device, address); - if (!deviceList.isEmpty()) { - return deviceList.itemAt(0); - } - deviceList = mHwModules[i]->mDeclaredDevices.getDevicesFromType(device); - if (!deviceList.isEmpty()) { - return deviceList.itemAt(0); - } - } - - sp<DeviceDescriptor> devDesc = new DeviceDescriptor(String8(""), device); - devDesc->mAddress = address; - return devDesc; + return deviceVector->getDeviceConnectionState(devDesc); } void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs) @@ -538,10 +348,11 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs AUDIO_OUTPUT_FLAG_NONE, AUDIO_FORMAT_INVALID); if (output != AUDIO_IO_HANDLE_NONE) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } @@ -550,8 +361,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs ALOGW_IF(status != NO_ERROR, "updateCallRouting() error %d creating RX audio patch", status); if (status == NO_ERROR) { - mCallRxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); + mCallRxPatch = new AudioPatch(&patch, mUidCached); mCallRxPatch->mAfPatchHandle = afPatchHandle; mCallRxPatch->mUid = mUidCached; } @@ -585,6 +395,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs ALOG_ASSERT(!outputDesc->isDuplicated(), "updateCallRouting() RX device output is duplicated"); outputDesc->toAudioPortConfig(&patch.sources[1]); + patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; patch.num_sources = 2; } @@ -593,8 +404,7 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs ALOGW_IF(status != NO_ERROR, "setPhoneState() error %d creating TX audio patch", status); if (status == NO_ERROR) { - mCallTxPatch = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); + mCallTxPatch = new AudioPatch(&patch, mUidCached); mCallTxPatch->mAfPatchHandle = afPatchHandle; mCallTxPatch->mUid = mUidCached; } @@ -604,16 +414,14 @@ void AudioPolicyManager::updateCallRouting(audio_devices_t rxDevice, int delayMs void AudioPolicyManager::setPhoneState(audio_mode_t state) { ALOGV("setPhoneState() state %d", state); - if (state < 0 || state >= AUDIO_MODE_CNT) { - ALOGW("setPhoneState() invalid state %d", state); - return; - } + // store previous phone state for management of sonification strategy below + int oldState = mEngine->getPhoneState(); - if (state == mPhoneState ) { - ALOGW("setPhoneState() setting same state %d", state); + if (mEngine->setPhoneState(state) != NO_ERROR) { + ALOGW("setPhoneState() invalid or same state %d", state); return; } - + /// Opens: can these line be executed after the switch of volume curves??? // if leaving call state, handle special case of active streams // pertaining to sonification strategy see handleIncallSonification() if (isInCall()) { @@ -629,66 +437,42 @@ void AudioPolicyManager::setPhoneState(audio_mode_t state) mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } - // store previous phone state for management of sonification strategy below - int oldState = mPhoneState; - mPhoneState = state; - bool force = false; - - // are we entering or starting a call - if (!isStateInCall(oldState) && isStateInCall(state)) { - ALOGV(" Entering call in setPhoneState()"); - // 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++) { - mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = - 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++) { - mStreams[AUDIO_STREAM_DTMF].mVolumeCurve[j] = - sVolumeProfiles[AUDIO_STREAM_DTMF][j]; - } - } else if (isStateInCall(state) && (state != oldState)) { - ALOGV(" Switching between telephony and VoIP in setPhoneState()"); - // force routing command to audio hardware when switching between telephony and VoIP - // even if no device change is needed - force = true; - } + /** + * Switching to or from incall state or switching between telephony and VoIP lead to force + * routing command. + */ + bool force = ((is_state_in_call(oldState) != is_state_in_call(state)) + || (is_state_in_call(state) && (state != oldState))); // check for device and output changes triggered by new phone state checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); - sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); + sp<SwAudioOutputDescriptor> hwOutputDesc = mPrimaryOutput; int delayMs = 0; if (isStateInCall(state)) { nsecs_t sysTime = systemTime(); for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); // mute media and sonification strategies and delay device switch by the largest // latency of any output where either strategy is active. // This avoid sending the ring tone or music tail into the earpiece or headset. - if ((desc->isStrategyActive(STRATEGY_MEDIA, - SONIFICATION_HEADSET_MUSIC_DELAY, - sysTime) || - desc->isStrategyActive(STRATEGY_SONIFICATION, - SONIFICATION_HEADSET_MUSIC_DELAY, - sysTime)) && - (delayMs < (int)desc->mLatency*2)) { - delayMs = desc->mLatency*2; - } - setStrategyMute(STRATEGY_MEDIA, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_MEDIA, false, mOutputs.keyAt(i), MUTE_TIME_MS, + if ((isStrategyActive(desc, STRATEGY_MEDIA, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime) || + isStrategyActive(desc, STRATEGY_SONIFICATION, + SONIFICATION_HEADSET_MUSIC_DELAY, + sysTime)) && + (delayMs < (int)desc->latency()*2)) { + delayMs = desc->latency()*2; + } + setStrategyMute(STRATEGY_MEDIA, true, desc); + setStrategyMute(STRATEGY_MEDIA, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/)); - setStrategyMute(STRATEGY_SONIFICATION, true, mOutputs.keyAt(i)); - setStrategyMute(STRATEGY_SONIFICATION, false, mOutputs.keyAt(i), MUTE_TIME_MS, + setStrategyMute(STRATEGY_SONIFICATION, true, desc); + setStrategyMute(STRATEGY_SONIFICATION, false, desc, MUTE_TIME_MS, getDeviceForStrategy(STRATEGY_SONIFICATION, true /*fromCache*/)); } } @@ -738,103 +522,49 @@ void AudioPolicyManager::setPhoneState(audio_mode_t state) } } +audio_mode_t AudioPolicyManager::getPhoneState() { + return mEngine->getPhoneState(); +} + void AudioPolicyManager::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) { - ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mPhoneState); - - bool forceVolumeReeval = false; - switch(usage) { - case AUDIO_POLICY_FORCE_FOR_COMMUNICATION: - if (config != AUDIO_POLICY_FORCE_SPEAKER && config != AUDIO_POLICY_FORCE_BT_SCO && - config != AUDIO_POLICY_FORCE_NONE) { - ALOGW("setForceUse() invalid config %d for FOR_COMMUNICATION", config); - return; - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_MEDIA: - if (config != AUDIO_POLICY_FORCE_HEADPHONES && config != AUDIO_POLICY_FORCE_BT_A2DP && - config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_ANALOG_DOCK && - config != AUDIO_POLICY_FORCE_DIGITAL_DOCK && config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_NO_BT_A2DP && config != AUDIO_POLICY_FORCE_SPEAKER ) { - ALOGW("setForceUse() invalid config %d for FOR_MEDIA", config); - return; - } - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_RECORD: - if (config != AUDIO_POLICY_FORCE_BT_SCO && config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_NONE) { - ALOGW("setForceUse() invalid config %d for FOR_RECORD", config); - return; - } - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_DOCK: - if (config != AUDIO_POLICY_FORCE_NONE && config != AUDIO_POLICY_FORCE_BT_CAR_DOCK && - config != AUDIO_POLICY_FORCE_BT_DESK_DOCK && - config != AUDIO_POLICY_FORCE_WIRED_ACCESSORY && - config != AUDIO_POLICY_FORCE_ANALOG_DOCK && - config != AUDIO_POLICY_FORCE_DIGITAL_DOCK) { - ALOGW("setForceUse() invalid config %d for FOR_DOCK", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_SYSTEM: - if (config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { - ALOGW("setForceUse() invalid config %d for FOR_SYSTEM", config); - } - forceVolumeReeval = true; - mForceUse[usage] = config; - break; - case AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO: - if (config != AUDIO_POLICY_FORCE_NONE && - config != AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED) { - ALOGW("setForceUse() invalid config %d forHDMI_SYSTEM_AUDIO", config); - } - mForceUse[usage] = config; - break; - default: - ALOGW("setForceUse() invalid usage %d", usage); - break; + ALOGV("setForceUse() usage %d, config %d, mPhoneState %d", usage, config, mEngine->getPhoneState()); + + if (mEngine->setForceUse(usage, config) != NO_ERROR) { + ALOGW("setForceUse() could not set force cfg %d for usage %d", config, usage); + return; } + bool forceVolumeReeval = (usage == AUDIO_POLICY_FORCE_FOR_COMMUNICATION) || + (usage == AUDIO_POLICY_FORCE_FOR_DOCK) || + (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM); // check for device and output changes triggered by new force usage checkA2dpSuspend(); checkOutputForAllStrategies(); updateDevicesAndOutputs(); - if (mPhoneState == AUDIO_MODE_IN_CALL) { + if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) { audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, true /*fromCache*/); updateCallRouting(newDevice); } for (size_t i = 0; i < mOutputs.size(); i++) { - audio_io_handle_t output = mOutputs.keyAt(i); - audio_devices_t newDevice = getNewOutputDevice(output, true /*fromCache*/); - if ((mPhoneState != AUDIO_MODE_IN_CALL) || (output != mPrimaryOutput)) { - setOutputDevice(output, newDevice, (newDevice != AUDIO_DEVICE_NONE)); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); + audio_devices_t newDevice = getNewOutputDevice(outputDesc, true /*fromCache*/); + if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (outputDesc != mPrimaryOutput)) { + setOutputDevice(outputDesc, newDevice, (newDevice != AUDIO_DEVICE_NONE)); } if (forceVolumeReeval && (newDevice != AUDIO_DEVICE_NONE)) { - applyStreamVolumes(output, newDevice, 0, true); + applyStreamVolumes(outputDesc, newDevice, 0, true); } } - audio_io_handle_t activeInput = getActiveInput(); + audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0) { setInputDevice(activeInput, getNewInputDevice(activeInput)); } } -audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) -{ - return mForceUse[usage]; -} - void AudioPolicyManager::setSystemProperty(const char* property, const char* value) { ALOGV("setSystemProperty() property %s, value %s", property, value); @@ -842,7 +572,7 @@ void AudioPolicyManager::setSystemProperty(const char* property, const char* val // 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, @@ -868,11 +598,11 @@ sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getProfileForDirectOutput( } audio_io_handle_t AudioPolicyManager::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) + uint32_t samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_output_flags_t flags, + const audio_offload_info_t *offloadInfo) { routing_strategy strategy = getStrategy(stream); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); @@ -892,6 +622,7 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + audio_port_handle_t selectedDeviceId, const audio_offload_info_t *offloadInfo) { audio_attributes_t attributes; @@ -910,45 +641,16 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, } stream_type_to_audio_attributes(*stream, &attributes); } - - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - sp<AudioOutputDescriptor> desc; - if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_PLAYERS) { - for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage == attributes.usage) || - (RULE_EXCLUDE_ATTRIBUTE_USAGE == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mUsage != attributes.usage)) { - desc = mPolicyMixes[i]->mOutput; - break; - } - if (strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && - strncmp(attributes.tags + strlen("addr="), - mPolicyMixes[i]->mMix.mRegistrationId.string(), - AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { - desc = mPolicyMixes[i]->mOutput; - break; - } - } - } else if (mPolicyMixes[i]->mMix.mMixType == MIX_TYPE_RECORDERS) { - if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE && - strncmp(attributes.tags, "addr=", strlen("addr=")) == 0 && - strncmp(attributes.tags + strlen("addr="), - mPolicyMixes[i]->mMix.mRegistrationId.string(), - AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0) { - desc = mPolicyMixes[i]->mOutput; - } - } - if (desc != 0) { - if (!audio_is_linear_pcm(format)) { - return BAD_VALUE; - } - desc->mPolicyMix = &mPolicyMixes[i]->mMix; - *stream = streamTypefromAttributesInt(&attributes); - *output = desc->mIoHandle; - ALOGV("getOutputForAttr() returns output %d", *output); - return NO_ERROR; + sp<SwAudioOutputDescriptor> desc; + if (mPolicyMixes.getOutputForAttr(attributes, desc) == NO_ERROR) { + ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr"); + if (!audio_is_linear_pcm(format)) { + return BAD_VALUE; } + *stream = streamTypefromAttributesInt(&attributes); + *output = desc->mIoHandle; + ALOGV("getOutputForAttr() returns output %d", *output); + return NO_ERROR; } if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) { ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE"); @@ -956,7 +658,7 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, } ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x", - attributes.usage, attributes.content_type, attributes.tags, attributes.flags); + attributes.usage, attributes.content_type, attributes.tags, attributes.flags); routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes); audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/); @@ -975,6 +677,17 @@ status_t AudioPolicyManager::getOutputForAttr(const audio_attributes_t *attr, if (*output == AUDIO_IO_HANDLE_NONE) { return INVALID_OPERATION; } + + // Explicit routing? + sp<DeviceDescriptor> deviceDesc; + + for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { + if (mAvailableOutputDevices[i]->getId() == selectedDeviceId) { + deviceDesc = mAvailableOutputDevices[i]; + break; + } + } + mOutputRoutes.addRoute(session, *stream, deviceDesc); return NO_ERROR; } @@ -999,7 +712,8 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (mTestOutputs[mCurOutput] == 0) { ALOGV("getOutput() opening test output"); - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL); + sp<AudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(NULL, + mpClientInterface); outputDesc->mDevice = mTestDevice; outputDesc->mLatency = mTestLatencyMs; outputDesc->mFlags = @@ -1066,7 +780,7 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( // in the background. if (((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) == 0) || - !isNonOffloadableEffectEnabled()) { + !mEffects.isNonOffloadableEffectEnabled()) { profile = getProfileForDirectOutput(device, samplingRate, format, @@ -1075,10 +789,10 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( } if (profile != 0) { - sp<AudioOutputDescriptor> outputDesc = NULL; + sp<SwAudioOutputDescriptor> outputDesc = NULL; for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); if (!desc->isDuplicated() && (profile == desc->mProfile)) { outputDesc = desc; // reuse direct output if currently open and configured with same parameters @@ -1095,7 +809,7 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (outputDesc != NULL) { closeOutput(outputDesc->mIoHandle); } - outputDesc = new AudioOutputDescriptor(profile); + outputDesc = new SwAudioOutputDescriptor(profile, mpClientInterface); outputDesc->mDevice = device; outputDesc->mLatency = 0; outputDesc->mFlags =(audio_output_flags_t) (outputDesc->mFlags | flags); @@ -1106,7 +820,7 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( if (offloadInfo != NULL) { config.offload_info = *offloadInfo; } - status = mpClientInterface->openOutput(profile->mModule->mHandle, + status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &outputDesc->mDevice, @@ -1130,6 +844,10 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( 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; @@ -1152,7 +870,6 @@ audio_io_handle_t AudioPolicyManager::getOutputForDevice( } non_direct_output: - // ignoring channel mask due to downmix capability in mixer // open a non direct output @@ -1170,7 +887,7 @@ non_direct_output: ALOGW_IF((output == 0), "getOutput() could not find output for stream %d, samplingRate %d," "format %d, channels %x, flags %x", stream, samplingRate, format, channelMask, flags); - ALOGV("getOutput() returns output %d", output); + ALOGV(" getOutputForDevice() returns output %d", output); return output; } @@ -1198,7 +915,7 @@ audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_h audio_io_handle_t outputPrimary = 0; for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); if (!outputDesc->isDuplicated()) { // if a valid format is specified, skip output if not compatible if (format != AUDIO_FORMAT_INVALID) { @@ -1237,18 +954,62 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, audio_stream_type_t stream, audio_session_t session) { - ALOGV("startOutput() output %d, stream %d, session %d", output, stream, session); + ALOGV("startOutput() output %d, stream %d, session %d", + output, stream, session); ssize_t index = mOutputs.indexOfKey(output); if (index < 0) { ALOGW("startOutput() unknown output %d", output); return BAD_VALUE; } + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + + audio_devices_t newDevice; + if (outputDesc->mPolicyMix != NULL) { + newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; + } else { + newDevice = AUDIO_DEVICE_NONE; + } + + uint32_t delayMs = 0; + + // Routing? + mOutputRoutes.incRouteActivity(session); + + status_t status = startSource(outputDesc, stream, newDevice, &delayMs); + + if (status != NO_ERROR) { + mOutputRoutes.decRouteActivity(session); + } + // Automatically enable the remote submix input when output is started on a re routing mix + // of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_AVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + + if (delayMs != 0) { + usleep(delayMs * 1000); + } + + return status; +} + +status_t AudioPolicyManager::startSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t *delayMs) +{ // cannot start playback of STREAM_TTS if any other output is being used uint32_t beaconMuteLatency = 0; + + *delayMs = 0; if (stream == AUDIO_STREAM_TTS) { ALOGV("\t found BEACON stream"); - if (isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { + if (mOutputs.isAnyOutputActive(AUDIO_STREAM_TTS /*streamToIgnore*/)) { return INVALID_OPERATION; } else { beaconMuteLatency = handleEventForBeacon(STARTING_BEACON); @@ -1258,8 +1019,6 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, beaconMuteLatency = handleEventForBeacon(STARTING_OUTPUT); } - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); - // increment usage count for this stream on the requested output: // NOTE that the usage count is the same for duplicated output and hardware output which is // necessary for a correct control of hardware output routing by startOutput() and stopOutput() @@ -1267,11 +1026,8 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, if (outputDesc->mRefCount[stream] == 1) { // starting an output being rerouted? - audio_devices_t newDevice; - if (outputDesc->mPolicyMix != NULL) { - newDevice = AUDIO_DEVICE_OUT_REMOTE_SUBMIX; - } else { - newDevice = getNewOutputDevice(output, false /*fromCache*/); + if (device == AUDIO_DEVICE_NONE) { + device = getNewOutputDevice(outputDesc, false /*fromCache*/); } routing_strategy strategy = getStrategy(stream); bool shouldWait = (strategy == STRATEGY_SONIFICATION) || @@ -1287,7 +1043,7 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, // In this case, the audio HAL must receive the new device selection so that it can // change the device currently selected by the other active output. if (outputDesc->sharesHwModuleWith(desc) && - desc->device() != newDevice) { + desc->device() != device) { force = true; } // wait for audio on other active outputs to be presented when starting @@ -1299,7 +1055,7 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, } } } - uint32_t muteWaitMs = setOutputDevice(output, newDevice, force); + uint32_t muteWaitMs = setOutputDevice(outputDesc, device, force); // handle special case for sonification while in call if (isInCall()) { @@ -1308,31 +1064,18 @@ status_t AudioPolicyManager::startOutput(audio_io_handle_t output, // apply volume rules for current stream and device if necessary checkAndSetVolume(stream, - mStreams[stream].getVolumeIndex(newDevice), - output, - newDevice); + mStreams.valueFor(stream).getVolumeIndex(device), + outputDesc, + device); // update the outputs if starting an output with a stream that can affect notification // routing handleNotificationRoutingForStream(stream); - // Automatically enable the remote submix input when output is started on a re routing mix - // of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(newDevice) && outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - outputDesc->mPolicyMix->mRegistrationId); - } - // force reevaluating accessibility routing when ringtone or alarm starts if (strategy == STRATEGY_SONIFICATION) { mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY); } - - if (waitMs > muteWaitMs) { - usleep((waitMs - muteWaitMs) * 2 * 1000); - } } return NO_ERROR; } @@ -1349,8 +1092,32 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, return BAD_VALUE; } - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); + + if (outputDesc->mRefCount[stream] == 1) { + // Automatically disable the remote submix input when output is stopped on a + // re routing mix of type MIX_TYPE_RECORDERS + if (audio_is_remote_submix_device(outputDesc->mDevice) && + outputDesc->mPolicyMix != NULL && + outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { + setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, + AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, + outputDesc->mPolicyMix->mRegistrationId, + "remote-submix"); + } + } + + // Routing? + if (outputDesc->mRefCount[stream] > 0) { + mOutputRoutes.decRouteActivity(session); + } + + return stopSource(outputDesc, stream); +} +status_t AudioPolicyManager::stopSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream) +{ // always handle stream stop, check which stream type is stopping handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT); @@ -1362,40 +1129,31 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, if (outputDesc->mRefCount[stream] > 0) { // decrement usage count of this stream on the output outputDesc->changeRefCount(stream, -1); + // store time at which the stream was stopped - see isStreamActive() if (outputDesc->mRefCount[stream] == 0) { - // Automatically disable the remote submix input when output is stopped on a - // re routing mix of type MIX_TYPE_RECORDERS - if (audio_is_remote_submix_device(outputDesc->mDevice) && - outputDesc->mPolicyMix != NULL && - outputDesc->mPolicyMix->mMixType == MIX_TYPE_RECORDERS) { - setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, - AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - outputDesc->mPolicyMix->mRegistrationId); - } - outputDesc->mStopTime[stream] = systemTime(); - audio_devices_t newDevice = getNewOutputDevice(output, false /*fromCache*/); + audio_devices_t newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/); // delay the device switch by twice the latency because stopOutput() is executed when // the track stop() command is received and at that time the audio track buffer can // still contain data that needs to be drained. The latency only covers the audio HAL // and kernel buffers. Also the latency does not always include additional delay in the // audio path (audio DSP, CODEC ...) - setOutputDevice(output, newDevice, false, outputDesc->mLatency*2); + setOutputDevice(outputDesc, newDevice, false, outputDesc->latency()*2); // force restoring the device selection on other active outputs if it differs from the // one being selected for this output for (size_t i = 0; i < mOutputs.size(); i++) { audio_io_handle_t curOutput = mOutputs.keyAt(i); sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); - if (curOutput != output && + if (desc != outputDesc && desc->isActive() && outputDesc->sharesHwModuleWith(desc) && (newDevice != desc->device())) { - setOutputDevice(curOutput, - getNewOutputDevice(curOutput, false /*fromCache*/), + setOutputDevice(desc, + getNewOutputDevice(desc, false /*fromCache*/), true, - outputDesc->mLatency*2); + outputDesc->latency()*2); } } // update the outputs if stopping one with a stream that can affect notification routing @@ -1403,7 +1161,7 @@ status_t AudioPolicyManager::stopOutput(audio_io_handle_t output, } return NO_ERROR; } else { - ALOGW("stopOutput() refcount is already 0 for output %d", output); + ALOGW("stopOutput() refcount is already 0"); return INVALID_OPERATION; } } @@ -1425,14 +1183,17 @@ void AudioPolicyManager::releaseOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); if (outputDesc->isActive()) { mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); + removeOutput(output); mTestOutputs[testIndex] = 0; } return; } #endif //AUDIO_POLICY_TEST - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(index); + // Routing + mOutputRoutes.removeRoute(session); + + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index); if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) { if (desc->mDirectOpenCount <= 0) { ALOGW("releaseOutput() invalid open count %d for output %d", @@ -1444,8 +1205,9 @@ void AudioPolicyManager::releaseOutput(audio_io_handle_t output, // If effects where present on the output, audioflinger moved them to the primary // output by default: move them back to the appropriate output. audio_io_handle_t dstOutput = getOutputForEffect(); - if (dstOutput != mPrimaryOutput) { - mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, mPrimaryOutput, dstOutput); + if (dstOutput != mPrimaryOutput->mIoHandle) { + mpClientInterface->moveEffects(AUDIO_SESSION_OUTPUT_MIX, + mPrimaryOutput->mIoHandle, dstOutput); } mpClientInterface->onAudioPortListUpdate(); } @@ -1483,19 +1245,13 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, if (inputSource == AUDIO_SOURCE_REMOTE_SUBMIX && strncmp(attr->tags, "addr=", strlen("addr=")) == 0) { - device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; - address = String8(attr->tags + strlen("addr=")); - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index < 0) { - ALOGW("getInputForAttr() no policy for address %s", address.string()); - return BAD_VALUE; - } - if (mPolicyMixes[index]->mMix.mMixType != MIX_TYPE_PLAYERS) { - ALOGW("getInputForAttr() bad policy mix type for address %s", address.string()); - return BAD_VALUE; + status_t ret = mPolicyMixes.getInputMixForAttr(*attr, &policyMix); + if (ret != NO_ERROR) { + return ret; } - policyMix = &mPolicyMixes[index]->mMix; *inputType = API_INPUT_MIX_EXT_POLICY_REROUTE; + device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; + address = String8(attr->tags + strlen("addr=")); } else { device = getDeviceAndMixForInputSource(inputSource, &policyMix); if (device == AUDIO_DEVICE_NONE) { @@ -1565,8 +1321,8 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, } } - if (profile->mModule->mHandle == 0) { - ALOGE("getInputForAttr(): HW module %s not opened", profile->mModule->mName); + if (profile->getModuleHandle() == 0) { + ALOGE("getInputForAttr(): HW module %s not opened", profile->getModuleName()); return NO_INIT; } @@ -1575,7 +1331,7 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, config.channel_mask = channelMask; config.format = format; - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + status_t status = mpClientInterface->openInput(profile->getModuleHandle(), input, &config, &device, @@ -1608,7 +1364,7 @@ status_t AudioPolicyManager::getInputForAttr(const audio_attributes_t *attr, inputDesc->mIsSoundTrigger = isSoundTrigger; inputDesc->mPolicyMix = policyMix; - ALOGV("getInputForAttr() returns input type = %d", inputType); + ALOGV("getInputForAttr() returns input type = %d", *inputType); addInput(*input, inputDesc); mpClientInterface->onAudioPortListUpdate(); @@ -1633,10 +1389,10 @@ status_t AudioPolicyManager::startInput(audio_io_handle_t input, } // virtual input devices are compatible with other input devices - if (!isVirtualInputDevice(inputDesc->mDevice)) { + if (!is_virtual_input_device(inputDesc->mDevice)) { // for a non-virtual input device, check if there is another (non-virtual) active input - audio_io_handle_t activeInput = getActiveInput(); + audio_io_handle_t activeInput = mInputs.getActiveInput(); if (activeInput != 0 && activeInput != input) { // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed, @@ -1654,7 +1410,7 @@ status_t AudioPolicyManager::startInput(audio_io_handle_t input, } if (inputDesc->mRefCount == 0) { - if (activeInputsCount() == 0) { + if (mInputs.activeInputsCount() == 0) { SoundTrigger::setCaptureState(true); } setInputDevice(input, getNewInputDevice(input), true /* force */); @@ -1672,7 +1428,7 @@ status_t AudioPolicyManager::startInput(audio_io_handle_t input, if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address); + address, "remote-submix"); } } } @@ -1720,13 +1476,13 @@ status_t AudioPolicyManager::stopInput(audio_io_handle_t input, if (address != "") { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address); + address, "remote-submix"); } } resetInputDevice(input); - if (activeInputsCount() == 0) { + if (mInputs.activeInputsCount() == 0) { SoundTrigger::setCaptureState(false); } } @@ -1793,25 +1549,20 @@ void AudioPolicyManager::initStreamVolume(audio_stream_type_t stream, int indexMax) { ALOGV("initStreamVolume() stream %d, min %d, max %d", stream , indexMin, indexMax); - if (indexMin < 0 || indexMin >= indexMax) { - ALOGW("initStreamVolume() invalid index limits for stream %d, min %d, max %d", stream , indexMin, indexMax); - return; - } - mStreams[stream].mIndexMin = indexMin; - mStreams[stream].mIndexMax = indexMax; + mEngine->initStreamVolume(stream, indexMin, indexMax); //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now if (stream == AUDIO_STREAM_MUSIC) { - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMin = indexMin; - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexMax = indexMax; + mEngine->initStreamVolume(AUDIO_STREAM_ACCESSIBILITY, indexMin, indexMax); } } status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, - int index, - audio_devices_t device) + int index, + audio_devices_t device) { - if ((index < mStreams[stream].mIndexMin) || (index > mStreams[stream].mIndexMax)) { + if ((index < mStreams.valueFor(stream).getVolumeIndexMin()) || + (index > mStreams.valueFor(stream).getVolumeIndexMax())) { return BAD_VALUE; } if (!audio_is_output_device(device)) { @@ -1819,7 +1570,7 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, } // Force max volume if stream cannot be muted - if (!mStreams[stream].mCanBeMuted) index = mStreams[stream].mIndexMax; + if (!mStreams.canBeMuted(stream)) index = mStreams.valueFor(stream).getVolumeIndexMax(); ALOGV("setStreamVolumeIndex() stream %d, device %04x, index %d", stream, device, index); @@ -1827,9 +1578,9 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, // if device is AUDIO_DEVICE_OUT_DEFAULT set default value and // clear all device specific values if (device == AUDIO_DEVICE_OUT_DEFAULT) { - mStreams[stream].mIndexCur.clear(); + mStreams.clearCurrentVolumeIndex(stream); } - mStreams[stream].mIndexCur.add(device, index); + mStreams.addCurrentVolumeIndex(stream, device, index); // update volume on all outputs whose current device is also selected by the same // strategy as the device specified by the caller @@ -1839,7 +1590,7 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, //FIXME: AUDIO_STREAM_ACCESSIBILITY volume follows AUDIO_STREAM_MUSIC for now audio_devices_t accessibilityDevice = AUDIO_DEVICE_NONE; if (stream == AUDIO_STREAM_MUSIC) { - mStreams[AUDIO_STREAM_ACCESSIBILITY].mIndexCur.add(device, index); + mStreams.addCurrentVolumeIndex(AUDIO_STREAM_ACCESSIBILITY, device, index); accessibilityDevice = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, true /*fromCache*/); } if ((device != AUDIO_DEVICE_OUT_DEFAULT) && @@ -1848,17 +1599,17 @@ status_t AudioPolicyManager::setStreamVolumeIndex(audio_stream_type_t stream, } status_t status = NO_ERROR; for (size_t i = 0; i < mOutputs.size(); i++) { - audio_devices_t curDevice = - getDeviceForVolume(mOutputs.valueAt(i)->device()); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); + audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device()); if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & strategyDevice) != 0)) { - status_t volStatus = checkAndSetVolume(stream, index, mOutputs.keyAt(i), curDevice); + status_t volStatus = checkAndSetVolume(stream, index, desc, curDevice); if (volStatus != NO_ERROR) { status = volStatus; } } if ((device == AUDIO_DEVICE_OUT_DEFAULT) || ((curDevice & accessibilityDevice) != 0)) { status_t volStatus = checkAndSetVolume(AUDIO_STREAM_ACCESSIBILITY, - index, mOutputs.keyAt(i), curDevice); + index, desc, curDevice); } } return status; @@ -1879,9 +1630,9 @@ status_t AudioPolicyManager::getStreamVolumeIndex(audio_stream_type_t stream, if (device == AUDIO_DEVICE_OUT_DEFAULT) { device = getDeviceForStrategy(getStrategy(stream), true /*fromCache*/); } - device = getDeviceForVolume(device); + device = Volume::getDeviceForVolume(device); - *index = mStreams[stream].getVolumeIndex(device); + *index = mStreams.valueFor(stream).getVolumeIndex(device); ALOGV("getStreamVolumeIndex() stream %d device %08x index %d", stream, device, *index); return NO_ERROR; } @@ -1905,7 +1656,7 @@ audio_io_handle_t AudioPolicyManager::selectOutputForEffects( audio_io_handle_t outputDeepBuffer = 0; for (size_t i = 0; i < outputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(outputs[i]); ALOGV("selectOutputForEffects outputs[%zu] flags %x", i, desc->mFlags); if ((desc->mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0) { outputOffloaded = outputs[i]; @@ -1956,137 +1707,17 @@ status_t AudioPolicyManager::registerEffect(const effect_descriptor_t *desc, return INVALID_OPERATION; } } - - if (mTotalEffectsMemory + desc->memoryUsage > getMaxEffectsMemory()) { - ALOGW("registerEffect() memory limit exceeded for Fx %s, Memory %d KB", - desc->name, desc->memoryUsage); - return INVALID_OPERATION; - } - mTotalEffectsMemory += desc->memoryUsage; - ALOGV("registerEffect() effect %s, io %d, strategy %d session %d id %d", - desc->name, io, strategy, session, id); - ALOGV("registerEffect() memory %d, total memory %d", desc->memoryUsage, mTotalEffectsMemory); - - sp<EffectDescriptor> effectDesc = new EffectDescriptor(); - memcpy (&effectDesc->mDesc, desc, sizeof(effect_descriptor_t)); - effectDesc->mIo = io; - effectDesc->mStrategy = (routing_strategy)strategy; - effectDesc->mSession = session; - effectDesc->mEnabled = false; - - mEffects.add(id, effectDesc); - - return NO_ERROR; -} - -status_t AudioPolicyManager::unregisterEffect(int id) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - ALOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - sp<EffectDescriptor> effectDesc = mEffects.valueAt(index); - - setEffectEnabled(effectDesc, false); - - if (mTotalEffectsMemory < effectDesc->mDesc.memoryUsage) { - ALOGW("unregisterEffect() memory %d too big for total %d", - effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); - effectDesc->mDesc.memoryUsage = mTotalEffectsMemory; - } - mTotalEffectsMemory -= effectDesc->mDesc.memoryUsage; - ALOGV("unregisterEffect() effect %s, ID %d, memory %d total memory %d", - effectDesc->mDesc.name, id, effectDesc->mDesc.memoryUsage, mTotalEffectsMemory); - - mEffects.removeItem(id); - - return NO_ERROR; -} - -status_t AudioPolicyManager::setEffectEnabled(int id, bool enabled) -{ - ssize_t index = mEffects.indexOfKey(id); - if (index < 0) { - ALOGW("unregisterEffect() unknown effect ID %d", id); - return INVALID_OPERATION; - } - - return setEffectEnabled(mEffects.valueAt(index), enabled); -} - -status_t AudioPolicyManager::setEffectEnabled(const sp<EffectDescriptor>& effectDesc, bool enabled) -{ - if (enabled == effectDesc->mEnabled) { - ALOGV("setEffectEnabled(%s) effect already %s", - enabled?"true":"false", enabled?"enabled":"disabled"); - return INVALID_OPERATION; - } - - if (enabled) { - if (mTotalEffectsCpuLoad + effectDesc->mDesc.cpuLoad > getMaxEffectsCpuLoad()) { - ALOGW("setEffectEnabled(true) CPU Load limit exceeded for Fx %s, CPU %f MIPS", - effectDesc->mDesc.name, (float)effectDesc->mDesc.cpuLoad/10); - return INVALID_OPERATION; - } - mTotalEffectsCpuLoad += effectDesc->mDesc.cpuLoad; - ALOGV("setEffectEnabled(true) total CPU %d", mTotalEffectsCpuLoad); - } else { - if (mTotalEffectsCpuLoad < effectDesc->mDesc.cpuLoad) { - ALOGW("setEffectEnabled(false) CPU load %d too high for total %d", - effectDesc->mDesc.cpuLoad, mTotalEffectsCpuLoad); - effectDesc->mDesc.cpuLoad = mTotalEffectsCpuLoad; - } - mTotalEffectsCpuLoad -= effectDesc->mDesc.cpuLoad; - ALOGV("setEffectEnabled(false) total CPU %d", mTotalEffectsCpuLoad); - } - effectDesc->mEnabled = enabled; - return NO_ERROR; -} - -bool AudioPolicyManager::isNonOffloadableEffectEnabled() -{ - for (size_t i = 0; i < mEffects.size(); i++) { - sp<EffectDescriptor> effectDesc = mEffects.valueAt(i); - if (effectDesc->mEnabled && (effectDesc->mStrategy == STRATEGY_MEDIA) && - ((effectDesc->mDesc.flags & EFFECT_FLAG_OFFLOAD_SUPPORTED) == 0)) { - ALOGV("isNonOffloadableEffectEnabled() non offloadable effect %s enabled on session %d", - effectDesc->mDesc.name, effectDesc->mSession); - return true; - } - } - return false; + return mEffects.registerEffect(desc, io, strategy, session, id); } bool AudioPolicyManager::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const { - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); - if (outputDesc->isStreamActive(stream, inPastMs, sysTime)) { - return true; - } - } - return false; + return mOutputs.isStreamActive(stream, inPastMs); } -bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, - uint32_t inPastMs) const +bool AudioPolicyManager::isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs) const { - nsecs_t sysTime = systemTime(); - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); - if (((outputDesc->device() & APM_AUDIO_OUT_DEVICE_REMOTE_ALL) != 0) && - outputDesc->isStreamActive(stream, inPastMs, sysTime)) { - // do not consider re routing (when the output is going to a dynamic policy) - // as "remote playback" - if (outputDesc->mPolicyMix == NULL) { - return true; - } - } - } - return false; + return mOutputs.isStreamActiveRemotely(stream, inPastMs); } bool AudioPolicyManager::isSourceActive(audio_source_t source) const @@ -2156,9 +1787,8 @@ status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes) for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index >= 0) { - ALOGE("registerPolicyMixes(): mix for address %s already registered", address.string()); + + if (mPolicyMixes.registerMix(address, mixes[i]) != NO_ERROR) { continue; } audio_config_t outputConfig = mixes[i].mFormat; @@ -2171,17 +1801,15 @@ status_t AudioPolicyManager::registerPolicyMixes(Vector<AudioMix> mixes) AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address); module->addInputProfile(address, &inputConfig, AUDIO_DEVICE_IN_REMOTE_SUBMIX, address); - sp<AudioPolicyMix> policyMix = new AudioPolicyMix(); - policyMix->mMix = mixes[i]; - mPolicyMixes.add(address, policyMix); + if (mixes[i].mMixType == MIX_TYPE_PLAYERS) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address.string()); + address.string(), "remote-submix"); } else { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_AVAILABLE, - address.string()); + address.string(), "remote-submix"); } } return NO_ERROR; @@ -2206,20 +1834,17 @@ status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes) for (size_t i = 0; i < mixes.size(); i++) { String8 address = mixes[i].mRegistrationId; - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index < 0) { - ALOGE("unregisterPolicyMixes(): mix for address %s not registered", address.string()); + + if (mPolicyMixes.unregisterMix(address) != NO_ERROR) { continue; } - mPolicyMixes.removeItemsAt(index); - if (getDeviceConnectionState(AUDIO_DEVICE_IN_REMOTE_SUBMIX, address.string()) == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { setDeviceConnectionStateInt(AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address.string()); + address.string(), "remote-submix"); } if (getDeviceConnectionState(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address.string()) == @@ -2227,7 +1852,7 @@ status_t AudioPolicyManager::unregisterPolicyMixes(Vector<AudioMix> mixes) { setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE, - address.string()); + address.string(), "remote-submix"); } module->removeOutputProfile(address); module->removeInputProfile(address); @@ -2245,89 +1870,34 @@ status_t AudioPolicyManager::dump(int fd) snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this); result.append(buffer); - snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput); + snprintf(buffer, SIZE, " Primary Output: %d\n", mPrimaryOutput->mIoHandle); result.append(buffer); - snprintf(buffer, SIZE, " Phone state: %d\n", mPhoneState); + snprintf(buffer, SIZE, " Phone state: %d\n", mEngine->getPhoneState()); result.append(buffer); snprintf(buffer, SIZE, " Force use for communications %d\n", - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)); result.append(buffer); - snprintf(buffer, SIZE, " Force use for media %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA]); + snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA)); result.append(buffer); - snprintf(buffer, SIZE, " Force use for record %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD]); + snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD)); result.append(buffer); - snprintf(buffer, SIZE, " Force use for dock %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK]); + snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK)); result.append(buffer); - snprintf(buffer, SIZE, " Force use for system %d\n", mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM]); + snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM)); result.append(buffer); snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n", - mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO]); - result.append(buffer); - - snprintf(buffer, SIZE, " Available output devices:\n"); + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO)); result.append(buffer); write(fd, result.string(), result.size()); - for (size_t i = 0; i < mAvailableOutputDevices.size(); i++) { - mAvailableOutputDevices[i]->dump(fd, 2, i); - } - snprintf(buffer, SIZE, "\n Available input devices:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { - mAvailableInputDevices[i]->dump(fd, 2, i); - } - - snprintf(buffer, SIZE, "\nHW Modules dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mHwModules.size(); i++) { - snprintf(buffer, SIZE, "- HW Module %zu:\n", i + 1); - write(fd, buffer, strlen(buffer)); - mHwModules[i]->dump(fd); - } - - snprintf(buffer, SIZE, "\nOutputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mOutputs.size(); i++) { - snprintf(buffer, SIZE, "- Output %d dump:\n", mOutputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mOutputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nInputs dump:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mInputs.size(); i++) { - snprintf(buffer, SIZE, "- Input %d dump:\n", mInputs.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mInputs.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nStreams dump:\n"); - write(fd, buffer, strlen(buffer)); - snprintf(buffer, SIZE, - " Stream Can be muted Index Min Index Max Index Cur [device : index]...\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < AUDIO_STREAM_CNT; i++) { - snprintf(buffer, SIZE, " %02zu ", i); - write(fd, buffer, strlen(buffer)); - mStreams[i].dump(fd); - } - - snprintf(buffer, SIZE, "\nTotal Effects CPU: %f MIPS, Total Effects memory: %d KB\n", - (float)mTotalEffectsCpuLoad/10, mTotalEffectsMemory); - write(fd, buffer, strlen(buffer)); - snprintf(buffer, SIZE, "Registered effects:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mEffects.size(); i++) { - snprintf(buffer, SIZE, "- Effect %d dump:\n", mEffects.keyAt(i)); - write(fd, buffer, strlen(buffer)); - mEffects.valueAt(i)->dump(fd); - } - - snprintf(buffer, SIZE, "\nAudio Patches:\n"); - write(fd, buffer, strlen(buffer)); - for (size_t i = 0; i < mAudioPatches.size(); i++) { - mAudioPatches[i]->dump(fd, 2, i); - } + mAvailableOutputDevices.dump(fd, String8("output")); + mAvailableInputDevices.dump(fd, String8("input")); + mHwModules.dump(fd); + mOutputs.dump(fd); + mInputs.dump(fd); + mStreams.dump(fd); + mEffects.dump(fd); + mAudioPatches.dump(fd); return NO_ERROR; } @@ -2384,7 +1954,7 @@ bool AudioPolicyManager::isOffloadSupported(const audio_offload_info_t& offloadI // FIXME: We should check the audio session here but we do not have it in this context. // This may prevent offloading in rare situations where effects are left active by apps // in the background. - if (isNonOffloadableEffectEnabled()) { + if (mEffects.isNonOffloadableEffectEnabled()) { return false; } @@ -2463,93 +2033,6 @@ status_t AudioPolicyManager::getAudioPort(struct audio_port *port __unused) return NO_ERROR; } -sp<AudioPolicyManager::AudioOutputDescriptor> AudioPolicyManager::getOutputFromId( - audio_port_handle_t id) const -{ - sp<AudioOutputDescriptor> outputDesc = NULL; - for (size_t i = 0; i < mOutputs.size(); i++) { - outputDesc = mOutputs.valueAt(i); - if (outputDesc->mId == id) { - break; - } - } - return outputDesc; -} - -sp<AudioPolicyManager::AudioInputDescriptor> AudioPolicyManager::getInputFromId( - audio_port_handle_t id) const -{ - sp<AudioInputDescriptor> inputDesc = NULL; - for (size_t i = 0; i < mInputs.size(); i++) { - inputDesc = mInputs.valueAt(i); - if (inputDesc->mId == id) { - break; - } - } - return inputDesc; -} - -sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleForDevice( - audio_devices_t device) const -{ - sp <HwModule> module; - - for (size_t i = 0; i < mHwModules.size(); i++) { - if (mHwModules[i]->mHandle == 0) { - continue; - } - if (audio_is_output_device(device)) { - for (size_t j = 0; j < mHwModules[i]->mOutputProfiles.size(); j++) - { - if (mHwModules[i]->mOutputProfiles[j]->mSupportedDevices.types() & device) { - return mHwModules[i]; - } - } - } else { - for (size_t j = 0; j < mHwModules[i]->mInputProfiles.size(); j++) { - if (mHwModules[i]->mInputProfiles[j]->mSupportedDevices.types() & - device & ~AUDIO_DEVICE_BIT_IN) { - return mHwModules[i]; - } - } - } - } - return module; -} - -sp <AudioPolicyManager::HwModule> AudioPolicyManager::getModuleFromName(const char *name) const -{ - sp <HwModule> module; - - for (size_t i = 0; i < mHwModules.size(); i++) - { - if (strcmp(mHwModules[i]->mName, name) == 0) { - return mHwModules[i]; - } - } - return module; -} - -audio_devices_t AudioPolicyManager::availablePrimaryOutputDevices() -{ - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); - audio_devices_t devices = outputDesc->mProfile->mSupportedDevices.types(); - return devices & mAvailableOutputDevices.types(); -} - -audio_devices_t AudioPolicyManager::availablePrimaryInputDevices() -{ - audio_module_handle_t primaryHandle = - mOutputs.valueFor(mPrimaryOutput)->mProfile->mModule->mHandle; - audio_devices_t devices = AUDIO_DEVICE_NONE; - for (size_t i = 0; i < mAvailableInputDevices.size(); i++) { - if (mAvailableInputDevices[i]->mModule->mHandle == primaryHandle) { - devices |= mAvailableInputDevices[i]->mDeviceType; - } - } - return devices; -} - status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, audio_patch_handle_t *handle, uid_t uid) @@ -2605,7 +2088,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, } if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("createAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; @@ -2634,15 +2117,14 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, return BAD_VALUE; } - if (!outputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, + if (!outputDesc->mProfile->isCompatibleProfile(devDesc->type(), devDesc->mAddress, patch->sources[0].sample_rate, - NULL, // updatedSamplingRate - patch->sources[0].format, - patch->sources[0].channel_mask, - AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { - ALOGV("createAudioPatch() profile not supported for device %08x", - devDesc->mDeviceType); + NULL, // updatedSamplingRate + patch->sources[0].format, + patch->sources[0].channel_mask, + AUDIO_OUTPUT_FLAG_NONE /*FIXME*/)) { + ALOGV("createAudioPatch() profile not supported for device %08x", devDesc->type()); return INVALID_OPERATION; } devices.add(devDesc); @@ -2654,7 +2136,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", devices.types(), outputDesc->mIoHandle); - setOutputDevice(outputDesc->mIoHandle, devices.types(), true, 0, handle); + setOutputDevice(outputDesc, devices.types(), true, 0, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { @@ -2674,7 +2156,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, if (patch->num_sinks > 1) { return INVALID_OPERATION; } - sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id); + sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { return BAD_VALUE; } @@ -2689,7 +2171,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, return BAD_VALUE; } - if (!inputDesc->mProfile->isCompatibleProfile(devDesc->mDeviceType, + if (!inputDesc->mProfile->isCompatibleProfile(devDesc->type(), devDesc->mAddress, patch->sinks[0].sample_rate, NULL, /*updatedSampleRate*/ @@ -2703,8 +2185,8 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, } // TODO: reconfigure output format and channels here ALOGV("createAudioPatch() setting device %08x on output %d", - devDesc->mDeviceType, inputDesc->mIoHandle); - setInputDevice(inputDesc->mIoHandle, devDesc->mDeviceType, true, handle); + devDesc->type(), inputDesc->mIoHandle); + setInputDevice(inputDesc->mIoHandle, devDesc->type(), true, handle); index = mAudioPatches.indexOfKey(*handle); if (index >= 0) { if (patchDesc != 0 && patchDesc != mAudioPatches.valueAt(index)) { @@ -2748,14 +2230,17 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, } sinkDeviceDesc->toAudioPortConfig(&newPatch.sinks[i], &patch->sinks[i]); - if (srcDeviceDesc->mModule != sinkDeviceDesc->mModule) { - // only one sink supported when connected devices across HW modules + // create a software bridge in PatchPanel if: + // - source and sink devices are on differnt HW modules OR + // - audio HAL version is < 3.0 + if ((srcDeviceDesc->getModuleHandle() != sinkDeviceDesc->getModuleHandle()) || + (srcDeviceDesc->mModule->mHalVersion < AUDIO_DEVICE_API_VERSION_3_0)) { + // support only one sink device for now to simplify output selection logic if (patch->num_sinks > 1) { return INVALID_OPERATION; } SortedVector<audio_io_handle_t> outputs = - getOutputsForDevice(sinkDeviceDesc->mDeviceType, - mOutputs); + getOutputsForDevice(sinkDeviceDesc->type(), mOutputs); // if the sink device is reachable via an opened output stream, request to go via // this output stream by adding a second source to the patch description audio_io_handle_t output = selectOutput(outputs, @@ -2767,6 +2252,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, return INVALID_OPERATION; } outputDesc->toAudioPortConfig(&newPatch.sources[1], &patch->sources[0]); + newPatch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH; newPatch.num_sources = 2; } } @@ -2785,8 +2271,7 @@ status_t AudioPolicyManager::createAudioPatch(const struct audio_patch *patch, status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &newPatch, uid); + patchDesc = new AudioPatch(&newPatch, uid); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = newPatch; @@ -2829,20 +2314,20 @@ status_t AudioPolicyManager::releaseAudioPatch(audio_patch_handle_t handle, struct audio_patch *patch = &patchDesc->mPatch; patchDesc->mUid = mUidCached; if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) { - sp<AudioOutputDescriptor> outputDesc = getOutputFromId(patch->sources[0].id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(patch->sources[0].id); if (outputDesc == NULL) { ALOGV("releaseAudioPatch() output not found for id %d", patch->sources[0].id); return BAD_VALUE; } - setOutputDevice(outputDesc->mIoHandle, - getNewOutputDevice(outputDesc->mIoHandle, true /*fromCache*/), + setOutputDevice(outputDesc, + getNewOutputDevice(outputDesc, true /*fromCache*/), true, 0, NULL); } else if (patch->sources[0].type == AUDIO_PORT_TYPE_DEVICE) { if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) { - sp<AudioInputDescriptor> inputDesc = getInputFromId(patch->sinks[0].id); + sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(patch->sinks[0].id); if (inputDesc == NULL) { ALOGV("releaseAudioPatch() input not found for id %d", patch->sinks[0].id); return BAD_VALUE; @@ -2872,30 +2357,11 @@ status_t AudioPolicyManager::listAudioPatches(unsigned int *num_patches, struct audio_patch *patches, unsigned int *generation) { - if (num_patches == NULL || (*num_patches != 0 && patches == NULL) || - generation == NULL) { + if (generation == NULL) { return BAD_VALUE; } - ALOGV("listAudioPatches() num_patches %d patches %p available patches %zu", - *num_patches, patches, mAudioPatches.size()); - if (patches == NULL) { - *num_patches = 0; - } - - size_t patchesWritten = 0; - size_t patchesMax = *num_patches; - for (size_t i = 0; - i < mAudioPatches.size() && patchesWritten < patchesMax; i++) { - patches[patchesWritten] = mAudioPatches[i]->mPatch; - patches[patchesWritten++].id = mAudioPatches[i]->mHandle; - ALOGV("listAudioPatches() patch %zu num_sources %d num_sinks %d", - i, mAudioPatches[i]->mPatch.num_sources, mAudioPatches[i]->mPatch.num_sinks); - } - *num_patches = mAudioPatches.size(); - *generation = curAudioPortGeneration(); - ALOGV("listAudioPatches() got %zu patches needed %d", patchesWritten, *num_patches); - return NO_ERROR; + return mAudioPatches.listAudioPatches(num_patches, patches); } status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config *config) @@ -2914,7 +2380,7 @@ status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config * sp<AudioPortConfig> audioPortConfig; if (config->type == AUDIO_PORT_TYPE_MIX) { if (config->role == AUDIO_PORT_ROLE_SOURCE) { - sp<AudioOutputDescriptor> outputDesc = getOutputFromId(config->id); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputFromId(config->id); if (outputDesc == NULL) { return BAD_VALUE; } @@ -2923,7 +2389,7 @@ status_t AudioPolicyManager::setAudioPortConfig(const struct audio_port_config * outputDesc->mIoHandle); audioPortConfig = outputDesc; } else if (config->role == AUDIO_PORT_ROLE_SINK) { - sp<AudioInputDescriptor> inputDesc = getInputFromId(config->id); + sp<AudioInputDescriptor> inputDesc = mInputs.getInputFromId(config->id); if (inputDesc == NULL) { return BAD_VALUE; } @@ -2980,63 +2446,12 @@ status_t AudioPolicyManager::acquireSoundTriggerSession(audio_session_t *session *ioHandle = (audio_io_handle_t)mpClientInterface->newAudioUniqueId(); *device = getDeviceAndMixForInputSource(AUDIO_SOURCE_HOTWORD); - mSoundTriggerSessions.add(*session, *ioHandle); - - return NO_ERROR; -} - -status_t AudioPolicyManager::releaseSoundTriggerSession(audio_session_t session) -{ - ssize_t index = mSoundTriggerSessions.indexOfKey(session); - if (index < 0) { - ALOGW("acquireSoundTriggerSession() session %d not registered", session); - return BAD_VALUE; - } - - mSoundTriggerSessions.removeItem(session); - return NO_ERROR; -} - -status_t AudioPolicyManager::addAudioPatch(audio_patch_handle_t handle, - const sp<AudioPatch>& patch) -{ - ssize_t index = mAudioPatches.indexOfKey(handle); - - if (index >= 0) { - ALOGW("addAudioPatch() patch %d already in", handle); - return ALREADY_EXISTS; - } - mAudioPatches.add(handle, patch); - ALOGV("addAudioPatch() handle %d af handle %d num_sources %d num_sinks %d source handle %d" - "sink handle %d", - handle, patch->mAfPatchHandle, patch->mPatch.num_sources, patch->mPatch.num_sinks, - patch->mPatch.sources[0].id, patch->mPatch.sinks[0].id); - return NO_ERROR; -} - -status_t AudioPolicyManager::removeAudioPatch(audio_patch_handle_t handle) -{ - ssize_t index = mAudioPatches.indexOfKey(handle); - - if (index < 0) { - ALOGW("removeAudioPatch() patch %d not in", handle); - return ALREADY_EXISTS; - } - ALOGV("removeAudioPatch() handle %d af handle %d", handle, - mAudioPatches.valueAt(index)->mAfPatchHandle); - mAudioPatches.removeItemsAt(index); - return NO_ERROR; + return mSoundTriggerSessions.acquireSession(*session, *ioHandle); } // ---------------------------------------------------------------------------- // AudioPolicyManager // ---------------------------------------------------------------------------- - -uint32_t AudioPolicyManager::nextUniqueId() -{ - return android_atomic_inc(&mNextUniqueId); -} - uint32_t AudioPolicyManager::nextAudioPortGeneration() { return android_atomic_inc(&mAudioPortGeneration); @@ -3047,35 +2462,47 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa #ifdef AUDIO_POLICY_TEST Thread(false), #endif //AUDIO_POLICY_TEST - mPrimaryOutput((audio_io_handle_t)0), - mPhoneState(AUDIO_MODE_NORMAL), mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f), - mTotalEffectsCpuLoad(0), mTotalEffectsMemory(0), mA2dpSuspended(false), - mSpeakerDrcEnabled(false), mNextUniqueId(1), + mSpeakerDrcEnabled(false), mAudioPortGeneration(1), mBeaconMuteRefCount(0), mBeaconPlayingRefCount(0), mBeaconMuted(false) { + audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance(); + if (!engineInstance) { + ALOGE("%s: Could not get an instance of policy engine", __FUNCTION__); + return; + } + // Retrieve the Policy Manager Interface + mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>(); + if (mEngine == NULL) { + ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__); + return; + } + mEngine->setObserver(this); + status_t status = mEngine->initCheck(); + ALOG_ASSERT(status == NO_ERROR, "Policy engine not initialized(err=%d)", status); + mUidCached = getuid(); mpClientInterface = clientInterface; - for (int i = 0; i < AUDIO_POLICY_FORCE_USE_CNT; i++) { - mForceUse[i] = AUDIO_POLICY_FORCE_NONE; - } - - mDefaultOutputDevice = new DeviceDescriptor(String8(""), AUDIO_DEVICE_OUT_SPEAKER); - if (loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE) != NO_ERROR) { - if (loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE) != NO_ERROR) { + mDefaultOutputDevice = new DeviceDescriptor(String8("Speaker"), AUDIO_DEVICE_OUT_SPEAKER); + if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_VENDOR_CONFIG_FILE, + mHwModules, mAvailableInputDevices, mAvailableOutputDevices, + mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) { + if (ConfigParsingUtils::loadAudioPolicyConfig(AUDIO_POLICY_CONFIG_FILE, + mHwModules, mAvailableInputDevices, mAvailableOutputDevices, + mDefaultOutputDevice, mSpeakerDrcEnabled) != NO_ERROR) { ALOGE("could not load audio policy configuration file, setting defaults"); defaultAudioPolicyConfig(); } } // mAvailableOutputDevices and mAvailableInputDevices now contain all attached devices - // must be done after reading the policy - initializeVolumeCurves(); + // must be done after reading the policy (since conditionned by Speaker Drc Enabling) + mEngine->initializeVolumeCurves(mSpeakerDrcEnabled); // open all output streams needed to access attached devices audio_devices_t outputDeviceTypes = mAvailableOutputDevices.types(); @@ -3103,13 +2530,13 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa continue; } audio_devices_t profileType = outProfile->mSupportedDevices.types(); - if ((profileType & mDefaultOutputDevice->mDeviceType) != AUDIO_DEVICE_NONE) { - profileType = mDefaultOutputDevice->mDeviceType; + if ((profileType & mDefaultOutputDevice->type()) != AUDIO_DEVICE_NONE) { + profileType = mDefaultOutputDevice->type(); } else { // chose first device present in mSupportedDevices also part of // outputDeviceTypes for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { - profileType = outProfile->mSupportedDevices[k]->mDeviceType; + profileType = outProfile->mSupportedDevices[k]->type(); if ((profileType & outputDeviceTypes) != 0) { break; } @@ -3118,7 +2545,8 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa if ((profileType & outputDeviceTypes) == 0) { continue; } - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(outProfile); + sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile, + mpClientInterface); outputDesc->mDevice = profileType; audio_config_t config = AUDIO_CONFIG_INITIALIZER; @@ -3126,7 +2554,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(outProfile->mModule->mHandle, + status_t status = mpClientInterface->openOutput(outProfile->getModuleHandle(), &output, &config, &outputDesc->mDevice, @@ -3144,21 +2572,20 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa outputDesc->mFormat = config.format; for (size_t k = 0; k < outProfile->mSupportedDevices.size(); k++) { - audio_devices_t type = outProfile->mSupportedDevices[k]->mDeviceType; + audio_devices_t type = outProfile->mSupportedDevices[k]->type(); ssize_t index = mAvailableOutputDevices.indexOf(outProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable - if ((index >= 0) && (mAvailableOutputDevices[index]->mId == 0)) { - mAvailableOutputDevices[index]->mId = nextUniqueId(); - mAvailableOutputDevices[index]->mModule = mHwModules[i]; + if (index >= 0 && !mAvailableOutputDevices[index]->isAttached()) { + mAvailableOutputDevices[index]->attach(mHwModules[i]); } } if (mPrimaryOutput == 0 && outProfile->mFlags & AUDIO_OUTPUT_FLAG_PRIMARY) { - mPrimaryOutput = output; + mPrimaryOutput = outputDesc; } addOutput(output, outputDesc); - setOutputDevice(output, + setOutputDevice(outputDesc, outputDesc->mDevice, true); } @@ -3177,7 +2604,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa // inputDeviceTypes audio_devices_t profileType = AUDIO_DEVICE_NONE; for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { - profileType = inProfile->mSupportedDevices[k]->mDeviceType; + profileType = inProfile->mSupportedDevices[k]->type(); if (profileType & inputDeviceTypes) { break; } @@ -3203,7 +2630,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa config.channel_mask = inputDesc->mChannelMask; config.format = inputDesc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(inProfile->mModule->mHandle, + status_t status = mpClientInterface->openInput(inProfile->getModuleHandle(), &input, &config, &inputDesc->mDevice, @@ -3213,13 +2640,12 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa if (status == NO_ERROR) { for (size_t k = 0; k < inProfile->mSupportedDevices.size(); k++) { - audio_devices_t type = inProfile->mSupportedDevices[k]->mDeviceType; + audio_devices_t type = inProfile->mSupportedDevices[k]->type(); ssize_t index = mAvailableInputDevices.indexOf(inProfile->mSupportedDevices[k]); // give a valid ID to an attached device once confirmed it is reachable - if ((index >= 0) && (mAvailableInputDevices[index]->mId == 0)) { - mAvailableInputDevices[index]->mId = nextUniqueId(); - mAvailableInputDevices[index]->mModule = mHwModules[i]; + if (index >= 0 && !mAvailableInputDevices[index]->isAttached()) { + mAvailableInputDevices[index]->attach(mHwModules[i]); } } mpClientInterface->closeInput(input); @@ -3232,24 +2658,30 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa } // make sure all attached devices have been allocated a unique ID for (size_t i = 0; i < mAvailableOutputDevices.size();) { - if (mAvailableOutputDevices[i]->mId == 0) { - ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->mDeviceType); + if (!mAvailableOutputDevices[i]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableOutputDevices[i]->type()); mAvailableOutputDevices.remove(mAvailableOutputDevices[i]); continue; } + // The device is now validated and can be appended to the available devices of the engine + mEngine->setDeviceConnectionState(mAvailableOutputDevices[i], + AUDIO_POLICY_DEVICE_STATE_AVAILABLE); i++; } for (size_t i = 0; i < mAvailableInputDevices.size();) { - if (mAvailableInputDevices[i]->mId == 0) { - ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->mDeviceType); + if (!mAvailableInputDevices[i]->isAttached()) { + ALOGW("Input device %08x unreachable", mAvailableInputDevices[i]->type()); mAvailableInputDevices.remove(mAvailableInputDevices[i]); continue; } + // The device is now validated and can be appended to the available devices of the engine + mEngine->setDeviceConnectionState(mAvailableInputDevices[i], + AUDIO_POLICY_DEVICE_STATE_AVAILABLE); i++; } // make sure default device is reachable if (mAvailableOutputDevices.indexOf(mDefaultOutputDevice) < 0) { - ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->mDeviceType); + ALOGE("Default device %08x is unreachable", mDefaultOutputDevice->type()); } ALOGE_IF((mPrimaryOutput == 0), "Failed to open primary output"); @@ -3260,7 +2692,7 @@ AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterfa if (mPrimaryOutput != 0) { AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); + mpClientInterface->setParameters(mPrimaryOutput->mIoHandle, outputCmd.toString()); mTestDevice = AUDIO_DEVICE_OUT_SPEAKER; mTestSamplingRate = 44100; @@ -3400,21 +2832,21 @@ bool AudioPolicyManager::threadLoop() if (param.get(String8("test_cmd_policy_reopen"), value) == NO_ERROR) { param.remove(String8("test_cmd_policy_reopen")); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); - mpClientInterface->closeOutput(mPrimaryOutput); - - audio_module_handle_t moduleHandle = outputDesc->mModule->mHandle; + mpClientInterface->closeOutput(mpClientInterface->closeOutput(mPrimaryOutput);); - mOutputs.removeItem(mPrimaryOutput); + audio_module_handle_t moduleHandle = mPrimaryOutput->getModuleHandle(); - sp<AudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL); + removeOutput(mPrimaryOutput->mIoHandle); + sp<SwAudioOutputDescriptor> outputDesc = new AudioOutputDescriptor(NULL, + mpClientInterface); outputDesc->mDevice = AUDIO_DEVICE_OUT_SPEAKER; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = outputDesc->mSamplingRate; config.channel_mask = outputDesc->mChannelMask; config.format = outputDesc->mFormat; + audio_io_handle_t handle; status_t status = mpClientInterface->openOutput(moduleHandle, - &mPrimaryOutput, + &handle, &config, &outputDesc->mDevice, String8(""), @@ -3428,10 +2860,11 @@ bool AudioPolicyManager::threadLoop() outputDesc->mSamplingRate = config.sample_rate; outputDesc->mChannelMask = config.channel_mask; outputDesc->mFormat = config.format; + mPrimaryOutput = outputDesc; AudioParameter outputCmd = AudioParameter(); outputCmd.addInt(String8("set_id"), 0); - mpClientInterface->setParameters(mPrimaryOutput, outputCmd.toString()); - addOutput(mPrimaryOutput, outputDesc); + mpClientInterface->setParameters(handle, outputCmd.toString()); + addOutput(handle, outputDesc); } } @@ -3463,23 +2896,26 @@ int AudioPolicyManager::testOutputIndex(audio_io_handle_t output) // --- -void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<AudioOutputDescriptor> outputDesc) +void AudioPolicyManager::addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc) { - outputDesc->mIoHandle = output; - outputDesc->mId = nextUniqueId(); + outputDesc->setIoHandle(output); mOutputs.add(output, outputDesc); nextAudioPortGeneration(); } +void AudioPolicyManager::removeOutput(audio_io_handle_t output) +{ + mOutputs.removeItem(output); +} + void AudioPolicyManager::addInput(audio_io_handle_t input, sp<AudioInputDescriptor> inputDesc) { - inputDesc->mIoHandle = input; - inputDesc->mId = nextUniqueId(); + inputDesc->setIoHandle(input); mInputs.add(input, inputDesc); nextAudioPortGeneration(); } -void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc /*in*/, +void AudioPolicyManager::findIoHandlesByAddress(sp<SwAudioOutputDescriptor> desc /*in*/, const audio_devices_t device /*in*/, const String8 address /*in*/, SortedVector<audio_io_handle_t>& outputs /*out*/) { @@ -3493,12 +2929,12 @@ void AudioPolicyManager::findIoHandlesByAddress(sp<AudioOutputDescriptor> desc / } status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> devDesc, - audio_policy_dev_state_t state, - SortedVector<audio_io_handle_t>& outputs, - const String8 address) + audio_policy_dev_state_t state, + SortedVector<audio_io_handle_t>& outputs, + const String8 address) { - audio_devices_t device = devDesc->mDeviceType; - sp<AudioOutputDescriptor> desc; + audio_devices_t device = devDesc->type(); + sp<SwAudioOutputDescriptor> desc; // erase all current sample rates, formats and channel masks devDesc->clearCapabilities(); @@ -3506,8 +2942,8 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de // first list already open outputs that can be routed to this device for (size_t i = 0; i < mOutputs.size(); i++) { desc = mOutputs.valueAt(i); - if (!desc->isDuplicated() && (desc->mProfile->mSupportedDevices.types() & device)) { - if (!deviceDistinguishesOnAddress(device)) { + if (!desc->isDuplicated() && (desc->supportedDevices() & device)) { + if (!device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): adding opened output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); } else { @@ -3527,7 +2963,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de { sp<IOProfile> profile = mHwModules[i]->mOutputProfiles[j]; if (profile->mSupportedDevices.types() & device) { - if (!deviceDistinguishesOnAddress(device) || + if (!device_distinguishes_on_address(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkOutputsForDevice(): adding profile %zu from module %zu", j, i); @@ -3565,7 +3001,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de ALOGV("opening output for device %08x with params %s profile %p", device, address.string(), profile.get()); - desc = new AudioOutputDescriptor(profile); + desc = new SwAudioOutputDescriptor(profile, mpClientInterface); desc->mDevice = device; audio_config_t config = AUDIO_CONFIG_INITIALIZER; config.sample_rate = desc->mSamplingRate; @@ -3575,7 +3011,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de config.offload_info.channel_mask = desc->mChannelMask; config.offload_info.format = desc->mFormat; audio_io_handle_t output = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openOutput(profile->mModule->mHandle, + status_t status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, @@ -3645,7 +3081,7 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de config.offload_info.sample_rate = config.sample_rate; config.offload_info.channel_mask = config.channel_mask; config.offload_info.format = config.format; - status = mpClientInterface->openOutput(profile->mModule->mHandle, + status = mpClientInterface->openOutput(profile->getModuleHandle(), &output, &config, &desc->mDevice, @@ -3663,45 +3099,46 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de if (output != AUDIO_IO_HANDLE_NONE) { addOutput(output, desc); - if (deviceDistinguishesOnAddress(device) && address != "0") { - ssize_t index = mPolicyMixes.indexOfKey(address); - if (index >= 0) { - mPolicyMixes[index]->mOutput = desc; - desc->mPolicyMix = &mPolicyMixes[index]->mMix; - } else { + if (device_distinguishes_on_address(device) && address != "0") { + sp<AudioPolicyMix> policyMix; + if (mPolicyMixes.getAudioPolicyMix(address, policyMix) != NO_ERROR) { ALOGE("checkOutputsForDevice() cannot find policy for address %s", address.string()); } + policyMix->setOutput(desc); + desc->mPolicyMix = policyMix->getMix(); + } else if ((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) == 0) { // no duplicated output for direct outputs and // outputs used by dynamic policy mixes audio_io_handle_t duplicatedOutput = AUDIO_IO_HANDLE_NONE; // set initial stream volume for device - applyStreamVolumes(output, device, 0, true); + applyStreamVolumes(desc, device, 0, true); //TODO: configure audio effect output stage here // open a duplicating output thread for the new output and the primary output - duplicatedOutput = mpClientInterface->openDuplicateOutput(output, - mPrimaryOutput); + duplicatedOutput = + mpClientInterface->openDuplicateOutput(output, + mPrimaryOutput->mIoHandle); if (duplicatedOutput != AUDIO_IO_HANDLE_NONE) { // add duplicated output descriptor - sp<AudioOutputDescriptor> dupOutputDesc = - new AudioOutputDescriptor(NULL); - dupOutputDesc->mOutput1 = mOutputs.valueFor(mPrimaryOutput); - dupOutputDesc->mOutput2 = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> dupOutputDesc = + new SwAudioOutputDescriptor(NULL, mpClientInterface); + dupOutputDesc->mOutput1 = mPrimaryOutput; + dupOutputDesc->mOutput2 = desc; dupOutputDesc->mSamplingRate = desc->mSamplingRate; dupOutputDesc->mFormat = desc->mFormat; dupOutputDesc->mChannelMask = desc->mChannelMask; dupOutputDesc->mLatency = desc->mLatency; addOutput(duplicatedOutput, dupOutputDesc); - applyStreamVolumes(duplicatedOutput, device, 0, true); + applyStreamVolumes(dupOutputDesc, device, 0, true); } else { ALOGW("checkOutputsForDevice() could not open dup output for %d and %d", - mPrimaryOutput, output); + mPrimaryOutput->mIoHandle, output); mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); + removeOutput(output); nextAudioPortGeneration(); output = AUDIO_IO_HANDLE_NONE; } @@ -3718,10 +3155,10 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de outputs.add(output); devDesc->importAudioPort(profile); - if (deviceDistinguishesOnAddress(device)) { + if (device_distinguishes_on_address(device)) { ALOGV("checkOutputsForDevice(): setOutputDevice(dev=0x%x, addr=%s)", device, address.string()); - setOutputDevice(output, device, true/*force*/, 0/*delay*/, + setOutputDevice(desc, device, true/*force*/, 0/*delay*/, NULL/*patch handle*/, address.string()); } ALOGV("checkOutputsForDevice(): adding output %d", output); @@ -3738,11 +3175,10 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de desc = mOutputs.valueAt(i); if (!desc->isDuplicated()) { // exact match on device - if (deviceDistinguishesOnAddress(device) && - (desc->mProfile->mSupportedDevices.types() == device)) { + if (device_distinguishes_on_address(device) && + (desc->supportedDevices() == device)) { findIoHandlesByAddress(desc, device, address, outputs); - } else if (!(desc->mProfile->mSupportedDevices.types() - & mAvailableOutputDevices.types())) { + } else if (!(desc->supportedDevices() & mAvailableOutputDevices.types())) { ALOGV("checkOutputsForDevice(): disconnecting adding output %d", mOutputs.keyAt(i)); outputs.add(mOutputs.keyAt(i)); @@ -3781,9 +3217,9 @@ status_t AudioPolicyManager::checkOutputsForDevice(const sp<DeviceDescriptor> de } status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, - audio_policy_dev_state_t state, - SortedVector<audio_io_handle_t>& inputs, - const String8 address) + audio_policy_dev_state_t state, + SortedVector<audio_io_handle_t>& inputs, + const String8 address) { sp<AudioInputDescriptor> desc; if (state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE) { @@ -3810,7 +3246,7 @@ status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, sp<IOProfile> profile = mHwModules[module_idx]->mInputProfiles[profile_index]; if (profile->mSupportedDevices.types() & (device & ~AUDIO_DEVICE_BIT_IN)) { - if (!deviceDistinguishesOnAddress(device) || + if (!device_distinguishes_on_address(device) || address == profile->mSupportedDevices[0]->mAddress) { profiles.add(profile); ALOGV("checkInputsForDevice(): adding profile %zu from module %zu", @@ -3850,7 +3286,7 @@ status_t AudioPolicyManager::checkInputsForDevice(audio_devices_t device, config.channel_mask = desc->mChannelMask; config.format = desc->mFormat; audio_io_handle_t input = AUDIO_IO_HANDLE_NONE; - status_t status = mpClientInterface->openInput(profile->mModule->mHandle, + status_t status = mpClientInterface->openInput(profile->getModuleHandle(), &input, &config, &desc->mDevice, @@ -3977,21 +3413,16 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output) { ALOGV("closeOutput(%d)", output); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); if (outputDesc == NULL) { ALOGW("closeOutput() unknown output %d", output); return; } - - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - if (mPolicyMixes[i]->mOutput == outputDesc) { - mPolicyMixes[i]->mOutput.clear(); - } - } + mPolicyMixes.closeOutput(outputDesc); // look for duplicated outputs connected to the output being removed. for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> dupOutputDesc = mOutputs.valueAt(i); if (dupOutputDesc->isDuplicated() && (dupOutputDesc->mOutput1 == outputDesc || dupOutputDesc->mOutput2 == outputDesc)) { @@ -4013,7 +3444,7 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output) ALOGV("closeOutput() closing also duplicated output %d", duplicatedOutput); mpClientInterface->closeOutput(duplicatedOutput); - mOutputs.removeItem(duplicatedOutput); + removeOutput(duplicatedOutput); } } @@ -4032,7 +3463,7 @@ void AudioPolicyManager::closeOutput(audio_io_handle_t output) mpClientInterface->setParameters(output, param.toString()); mpClientInterface->closeOutput(output); - mOutputs.removeItem(output); + removeOutput(output); mPreviousOutputs = mOutputs; } @@ -4060,8 +3491,9 @@ void AudioPolicyManager::closeInput(audio_io_handle_t input) mInputs.removeItem(input); } -SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_devices_t device, - DefaultKeyedVector<audio_io_handle_t, sp<AudioOutputDescriptor> > openOutputs) +SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice( + audio_devices_t device, + SwAudioOutputCollection openOutputs) { SortedVector<audio_io_handle_t> outputs; @@ -4078,7 +3510,7 @@ SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(audio_de } bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1, - SortedVector<audio_io_handle_t>& outputs2) + SortedVector<audio_io_handle_t>& outputs2) { if (outputs1.size() != outputs2.size()) { return false; @@ -4102,14 +3534,14 @@ void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) // associated with policies in the "before" and "after" output vectors ALOGVV("checkOutputForStrategy(): policy related outputs"); for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) { - const sp<AudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); + const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { srcOutputs.add(desc->mIoHandle); ALOGVV(" previous outputs: adding %d", desc->mIoHandle); } } for (size_t i = 0 ; i < mOutputs.size() ; i++) { - const sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); if (desc != 0 && desc->mPolicyMix != NULL) { dstOutputs.add(desc->mIoHandle); ALOGVV(" new outputs: adding %d", desc->mIoHandle); @@ -4121,10 +3553,10 @@ void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) strategy, srcOutputs[0], dstOutputs[0]); // mute strategy while moving tracks from one output to another for (size_t i = 0; i < srcOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]); - if (desc->isStrategyActive(strategy)) { - setStrategyMute(strategy, true, srcOutputs[i]); - setStrategyMute(strategy, false, srcOutputs[i], MUTE_TIME_MS, newDevice); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(srcOutputs[i]); + if (isStrategyActive(desc, strategy)) { + setStrategyMute(strategy, true, desc); + setStrategyMute(strategy, false, desc, MUTE_TIME_MS, newDevice); } } @@ -4161,10 +3593,10 @@ void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy) void AudioPolicyManager::checkOutputForAllStrategies() { - if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) + if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_PHONE); - if (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) + if (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) != AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) checkOutputForStrategy(STRATEGY_ENFORCED_AUDIBLE); checkOutputForStrategy(STRATEGY_SONIFICATION); checkOutputForStrategy(STRATEGY_SONIFICATION_RESPECTFUL); @@ -4174,21 +3606,9 @@ void AudioPolicyManager::checkOutputForAllStrategies() checkOutputForStrategy(STRATEGY_REROUTING); } -audio_io_handle_t AudioPolicyManager::getA2dpOutput() -{ - for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); - if (!outputDesc->isDuplicated() && outputDesc->device() & AUDIO_DEVICE_OUT_ALL_A2DP) { - return mOutputs.keyAt(i); - } - } - - return 0; -} - void AudioPolicyManager::checkA2dpSuspend() { - audio_io_handle_t a2dpOutput = getA2dpOutput(); + audio_io_handle_t a2dpOutput = mOutputs.getA2dpOutput(); if (a2dpOutput == 0) { mA2dpSuspended = false; return; @@ -4212,20 +3632,20 @@ void AudioPolicyManager::checkA2dpSuspend() // if (mA2dpSuspended) { if ((!isScoConnected || - ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] != AUDIO_POLICY_FORCE_BT_SCO))) && - ((mPhoneState != AUDIO_MODE_IN_CALL) && - (mPhoneState != AUDIO_MODE_RINGTONE))) { + ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) != AUDIO_POLICY_FORCE_BT_SCO) && + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) != AUDIO_POLICY_FORCE_BT_SCO))) && + ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) && + (mEngine->getPhoneState() != AUDIO_MODE_RINGTONE))) { mpClientInterface->restoreOutput(a2dpOutput); mA2dpSuspended = false; } } else { if ((isScoConnected && - ((mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO))) || - ((mPhoneState == AUDIO_MODE_IN_CALL) || - (mPhoneState == AUDIO_MODE_RINGTONE))) { + ((mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) || + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO))) || + ((mEngine->getPhoneState() == AUDIO_MODE_IN_CALL) || + (mEngine->getPhoneState() == AUDIO_MODE_RINGTONE))) { mpClientInterface->suspendOutput(a2dpOutput); mA2dpSuspended = true; @@ -4233,12 +3653,11 @@ void AudioPolicyManager::checkA2dpSuspend() } } -audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, bool fromCache) +audio_devices_t AudioPolicyManager::getNewOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, + bool fromCache) { audio_devices_t device = AUDIO_DEVICE_NONE; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); - ssize_t index = mAudioPatches.indexOfKey(outputDesc->mPatchHandle); if (index >= 0) { sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index); @@ -4268,27 +3687,27 @@ audio_devices_t AudioPolicyManager::getNewOutputDevice(audio_io_handle_t output, // use device for strategy DTMF // 9: the strategy for beacon, a.k.a. "transmitted through speaker" is active on the output: // use device for strategy t-t-s - if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE) && - mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { + if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE) && + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); } else if (isInCall() || - outputDesc->isStrategyActive(STRATEGY_PHONE)) { + isStrategyActive(outputDesc, STRATEGY_PHONE)) { device = getDeviceForStrategy(STRATEGY_PHONE, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_ENFORCED_AUDIBLE)) { + } else if (isStrategyActive(outputDesc, STRATEGY_ENFORCED_AUDIBLE)) { device = getDeviceForStrategy(STRATEGY_ENFORCED_AUDIBLE, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION)) { + } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_SONIFICATION_RESPECTFUL)) { + } else if (isStrategyActive(outputDesc, STRATEGY_SONIFICATION_RESPECTFUL)) { device = getDeviceForStrategy(STRATEGY_SONIFICATION_RESPECTFUL, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_ACCESSIBILITY)) { + } else if (isStrategyActive(outputDesc, STRATEGY_ACCESSIBILITY)) { device = getDeviceForStrategy(STRATEGY_ACCESSIBILITY, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_MEDIA)) { + } else if (isStrategyActive(outputDesc, STRATEGY_MEDIA)) { device = getDeviceForStrategy(STRATEGY_MEDIA, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_DTMF)) { + } else if (isStrategyActive(outputDesc, STRATEGY_DTMF)) { device = getDeviceForStrategy(STRATEGY_DTMF, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { + } else if (isStrategyActive(outputDesc, STRATEGY_TRANSMITTED_THROUGH_SPEAKER)) { device = getDeviceForStrategy(STRATEGY_TRANSMITTED_THROUGH_SPEAKER, fromCache); - } else if (outputDesc->isStrategyActive(STRATEGY_REROUTING)) { + } else if (isStrategyActive(outputDesc, STRATEGY_REROUTING)) { device = getDeviceForStrategy(STRATEGY_REROUTING, fromCache); } @@ -4328,12 +3747,12 @@ audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stre 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++) { sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(outputs[i]); - if (outputDesc->isStrategyActive(strategy)) { + if (isStrategyActive(outputDesc, strategy)) { devices = outputDesc->device(); break; } @@ -4349,39 +3768,10 @@ audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stre return devices; } -AudioPolicyManager::routing_strategy AudioPolicyManager::getStrategy( - audio_stream_type_t stream) { - +routing_strategy AudioPolicyManager::getStrategy(audio_stream_type_t stream) const +{ ALOG_ASSERT(stream != AUDIO_STREAM_PATCH,"getStrategy() called for AUDIO_STREAM_PATCH"); - - // stream to strategy mapping - switch (stream) { - case AUDIO_STREAM_VOICE_CALL: - case AUDIO_STREAM_BLUETOOTH_SCO: - return STRATEGY_PHONE; - case AUDIO_STREAM_RING: - case AUDIO_STREAM_ALARM: - return STRATEGY_SONIFICATION; - case AUDIO_STREAM_NOTIFICATION: - return STRATEGY_SONIFICATION_RESPECTFUL; - case AUDIO_STREAM_DTMF: - return STRATEGY_DTMF; - default: - ALOGE("unknown stream type %d", stream); - case AUDIO_STREAM_SYSTEM: - // NOTE: SYSTEM stream uses MEDIA strategy because muting music and switching outputs - // while key clicks are played produces a poor result - case AUDIO_STREAM_MUSIC: - return STRATEGY_MEDIA; - case AUDIO_STREAM_ENFORCED_AUDIBLE: - return STRATEGY_ENFORCED_AUDIBLE; - case AUDIO_STREAM_TTS: - return STRATEGY_TRANSMITTED_THROUGH_SPEAKER; - case AUDIO_STREAM_ACCESSIBILITY: - return STRATEGY_ACCESSIBILITY; - case AUDIO_STREAM_REROUTING: - return STRATEGY_REROUTING; - } + return mEngine->getStrategyForStream(stream); } uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) { @@ -4392,45 +3782,8 @@ uint32_t AudioPolicyManager::getStrategyForAttr(const audio_attributes_t *attr) if ((attr->flags & AUDIO_FLAG_AUDIBILITY_ENFORCED) == AUDIO_FLAG_AUDIBILITY_ENFORCED) { return (uint32_t) STRATEGY_ENFORCED_AUDIBLE; } - // usage to strategy mapping - switch (attr->usage) { - case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY: - if (isStreamActive(AUDIO_STREAM_RING) || isStreamActive(AUDIO_STREAM_ALARM)) { - return (uint32_t) STRATEGY_SONIFICATION; - } - if (isInCall()) { - return (uint32_t) STRATEGY_PHONE; - } - return (uint32_t) STRATEGY_ACCESSIBILITY; - - case AUDIO_USAGE_MEDIA: - case AUDIO_USAGE_GAME: - case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE: - case AUDIO_USAGE_ASSISTANCE_SONIFICATION: - return (uint32_t) STRATEGY_MEDIA; - - case AUDIO_USAGE_VOICE_COMMUNICATION: - return (uint32_t) STRATEGY_PHONE; - - case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING: - return (uint32_t) STRATEGY_DTMF; - - case AUDIO_USAGE_ALARM: - case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE: - return (uint32_t) STRATEGY_SONIFICATION; - - case AUDIO_USAGE_NOTIFICATION: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT: - case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED: - case AUDIO_USAGE_NOTIFICATION_EVENT: - return (uint32_t) STRATEGY_SONIFICATION_RESPECTFUL; - - case AUDIO_USAGE_UNKNOWN: - default: - return (uint32_t) STRATEGY_MEDIA; - } + return static_cast<uint32_t>(mEngine->getStrategyForUsage(attr->usage)); } void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t stream) { @@ -4444,21 +3797,6 @@ void AudioPolicyManager::handleNotificationRoutingForStream(audio_stream_type_t } } -bool AudioPolicyManager::isAnyOutputActive(audio_stream_type_t streamToIgnore) { - for (size_t s = 0 ; s < AUDIO_STREAM_CNT ; s++) { - if (s == (size_t) streamToIgnore) { - continue; - } - for (size_t i = 0; i < mOutputs.size(); i++) { - const sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(i); - if (outputDesc->mRefCount[s] != 0) { - return true; - } - } - } - return false; -} - uint32_t AudioPolicyManager::handleEventForBeacon(int event) { switch(event) { case STARTING_OUTPUT: @@ -4497,9 +3835,9 @@ uint32_t AudioPolicyManager::setBeaconMute(bool mute) { ALOGV("\t muting %d", mute); uint32_t maxLatency = 0; for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); + sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i); setStreamMute(AUDIO_STREAM_TTS, mute/*on*/, - desc->mIoHandle, + desc, 0 /*delay*/, AUDIO_DEVICE_NONE); const uint32_t latency = desc->latency() * 2; if (latency > maxLatency) { @@ -4513,292 +3851,29 @@ uint32_t AudioPolicyManager::setBeaconMute(bool mute) { } audio_devices_t AudioPolicyManager::getDeviceForStrategy(routing_strategy strategy, - bool fromCache) + bool fromCache) { - uint32_t device = AUDIO_DEVICE_NONE; + // Routing + // see if we have an explicit route + // scan the whole RouteMap, for each entry, convert the stream type to a strategy + // (getStrategy(stream)). + // if the strategy from the stream type in the RouteMap is the same as the argument above, + // and activity count is non-zero + // the device = the device from the descriptor in the RouteMap, and exit. + for (size_t routeIndex = 0; routeIndex < mOutputRoutes.size(); routeIndex++) { + sp<SessionRoute> route = mOutputRoutes.valueAt(routeIndex); + routing_strategy strat = getStrategy(route->mStreamType); + if (strat == strategy && route->mDeviceDescriptor != 0 /*&& route->mActivityCount != 0*/) { + return route->mDeviceDescriptor->type(); + } + } if (fromCache) { ALOGVV("getDeviceForStrategy() from cache strategy %d, device %x", strategy, mDeviceForStrategy[strategy]); return mDeviceForStrategy[strategy]; } - audio_devices_t availableOutputDeviceTypes = mAvailableOutputDevices.types(); - switch (strategy) { - - case STRATEGY_TRANSMITTED_THROUGH_SPEAKER: - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (!device) { - ALOGE("getDeviceForStrategy() no device found for "\ - "STRATEGY_TRANSMITTED_THROUGH_SPEAKER"); - } - break; - - case STRATEGY_SONIFICATION_RESPECTFUL: - if (isInCall()) { - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - } else if (isStreamActiveRemotely(AUDIO_STREAM_MUSIC, - SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { - // while media is playing on a remote device, use the the sonification behavior. - // Note that we test this usecase before testing if media is playing because - // the isStreamActive() method only informs about the activity of a stream, not - // if it's for local playback. Note also that we use the same delay between both tests - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - //user "safe" speaker if available instead of normal speaker to avoid triggering - //other acoustic safety mechanisms for notification - if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) - device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; - } else if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY)) { - // while media is playing (or has recently played), use the same device - device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); - } else { - // when media is not playing anymore, fall back on the sonification behavior - device = getDeviceForStrategy(STRATEGY_SONIFICATION, false /*fromCache*/); - //user "safe" speaker if available instead of normal speaker to avoid triggering - //other acoustic safety mechanisms for notification - if (device == AUDIO_DEVICE_OUT_SPEAKER && (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) - device = AUDIO_DEVICE_OUT_SPEAKER_SAFE; - } - - break; - - case STRATEGY_DTMF: - if (!isInCall()) { - // when off call, DTMF strategy follows the same rules as MEDIA strategy - device = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/); - break; - } - // when in call, DTMF and PHONE strategies follow the same rules - // FALL THROUGH - - case STRATEGY_PHONE: - // Force use of only devices on primary output if: - // - in call AND - // - cannot route from voice call RX OR - // - audio HAL version is < 3.0 and TX device is on the primary HW module - if (mPhoneState == AUDIO_MODE_IN_CALL) { - audio_devices_t txDevice = - getDeviceAndMixForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION); - sp<AudioOutputDescriptor> hwOutputDesc = mOutputs.valueFor(mPrimaryOutput); - if (((mAvailableInputDevices.types() & - AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) || - (((txDevice & availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN) != 0) && - (hwOutputDesc->getAudioPort()->mModule->mHalVersion < - AUDIO_DEVICE_API_VERSION_3_0))) { - availableOutputDeviceTypes = availablePrimaryOutputDevices(); - } - } - // for phone strategy, we first consider the forced use and then the available devices by order - // of priority - switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { - case AUDIO_POLICY_FORCE_BT_SCO: - if (!isInCall() || strategy != STRATEGY_DTMF) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_SCO; - if (device) break; - // if SCO device is requested but no SCO device is available, fall back to default case - // FALL THROUGH - - default: // FORCE_NONE - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP - if (!isInCall() && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - if (device) break; - if (mPhoneState != AUDIO_MODE_IN_CALL) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_EARPIECE; - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE"); - } - break; - - case AUDIO_POLICY_FORCE_SPEAKER: - // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to - // A2DP speaker when forcing to speaker output - if (!isInCall() && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - if (device) break; - } - if (!isInCall()) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - if (device) break; - } - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; - if (device) break; - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_PHONE, FORCE_SPEAKER"); - } - break; - } - break; - - case STRATEGY_SONIFICATION: - - // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by - // handleIncallSonification(). - if (isInCall()) { - device = getDeviceForStrategy(STRATEGY_PHONE, false /*fromCache*/); - break; - } - // FALL THROUGH - - case STRATEGY_ENFORCED_AUDIBLE: - // strategy STRATEGY_ENFORCED_AUDIBLE uses same routing policy as STRATEGY_SONIFICATION - // except: - // - when in call where it doesn't default to STRATEGY_PHONE behavior - // - in countries where not enforced in which case it follows STRATEGY_MEDIA - - if ((strategy == STRATEGY_SONIFICATION) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) { - device = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() speaker device not found for STRATEGY_SONIFICATION"); - } - } - // The second device used for sonification is the same as the device used by media strategy - // FALL THROUGH - - // FIXME: STRATEGY_ACCESSIBILITY and STRATEGY_REROUTING follow STRATEGY_MEDIA for now - case STRATEGY_ACCESSIBILITY: - if (strategy == STRATEGY_ACCESSIBILITY) { - // do not route accessibility prompts to a digital output currently configured with a - // compressed format as they would likely not be mixed and dropped. - for (size_t i = 0; i < mOutputs.size(); i++) { - sp<AudioOutputDescriptor> desc = mOutputs.valueAt(i); - audio_devices_t devices = desc->device() & - (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC); - if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) && - devices != AUDIO_DEVICE_NONE) { - availableOutputDeviceTypes = availableOutputDeviceTypes & ~devices; - } - } - } - // FALL THROUGH - - case STRATEGY_REROUTING: - case STRATEGY_MEDIA: { - uint32_t device2 = AUDIO_DEVICE_NONE; - if (strategy != STRATEGY_SONIFICATION) { - // no sonification on remote submix (e.g. WFD) - if (mAvailableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0")) != 0) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_REMOTE_SUBMIX; - } - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] != AUDIO_POLICY_FORCE_NO_BT_A2DP) && - (getA2dpOutput() != 0)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP; - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER; - } - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_MEDIA] == AUDIO_POLICY_FORCE_SPEAKER)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; - } - if ((device2 == AUDIO_DEVICE_NONE)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_LINE; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_WIRED_HEADSET; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_ACCESSORY; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_USB_DEVICE; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET; - } - if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) { - // no sonification on aux digital (e.g. HDMI) - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_DIGITAL; - } - if ((device2 == AUDIO_DEVICE_NONE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_DOCK] == AUDIO_POLICY_FORCE_ANALOG_DOCK)) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET; - } - if (device2 == AUDIO_DEVICE_NONE) { - device2 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPEAKER; - } - int device3 = AUDIO_DEVICE_NONE; - if (strategy == STRATEGY_MEDIA) { - // ARC, SPDIF and AUX_LINE can co-exist with others. - device3 = availableOutputDeviceTypes & AUDIO_DEVICE_OUT_HDMI_ARC; - device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_SPDIF); - device3 |= (availableOutputDeviceTypes & AUDIO_DEVICE_OUT_AUX_LINE); - } - - device2 |= device3; - // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or - // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise - device |= device2; - - // If hdmi system audio mode is on, remove speaker out of output list. - if ((strategy == STRATEGY_MEDIA) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO] == - AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) { - device &= ~AUDIO_DEVICE_OUT_SPEAKER; - } - - if (device) break; - device = mDefaultOutputDevice->mDeviceType; - if (device == AUDIO_DEVICE_NONE) { - ALOGE("getDeviceForStrategy() no device found for STRATEGY_MEDIA"); - } - } break; - - default: - ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy); - break; - } - - ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device); - return device; + return mEngine->getDeviceForStrategy(strategy); } void AudioPolicyManager::updateDevicesAndOutputs() @@ -4826,7 +3901,7 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> for (size_t i = 0; i < NUM_STRATEGIES; i++) { audio_devices_t curDevice = getDeviceForStrategy((routing_strategy)i, false /*fromCache*/); - curDevice = curDevice & outputDesc->mProfile->mSupportedDevices.types(); + curDevice = curDevice & outputDesc->supportedDevices(); bool mute = shouldMute && (curDevice & device) && (curDevice != device); bool doMute = false; @@ -4845,11 +3920,10 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> == AUDIO_DEVICE_NONE) { continue; } - audio_io_handle_t curOutput = mOutputs.keyAt(j); - ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x) on output %d", - mute ? "muting" : "unmuting", i, curDevice, curOutput); - setStrategyMute((routing_strategy)i, mute, curOutput, mute ? 0 : delayMs); - if (desc->isStrategyActive((routing_strategy)i)) { + ALOGVV("checkDeviceMuteStrategies() %s strategy %d (curDevice %04x)", + mute ? "muting" : "unmuting", i, curDevice); + setStrategyMute((routing_strategy)i, mute, desc, mute ? 0 : delayMs); + if (isStrategyActive(desc, (routing_strategy)i)) { if (mute) { // FIXME: should not need to double latency if volume could be applied // immediately by the audioflinger mixer. We must account for the delay @@ -4872,10 +3946,10 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> muteWaitMs = outputDesc->latency() * 2; } for (size_t i = 0; i < NUM_STRATEGIES; i++) { - if (outputDesc->isStrategyActive((routing_strategy)i)) { - setStrategyMute((routing_strategy)i, true, outputDesc->mIoHandle); + if (isStrategyActive(outputDesc, (routing_strategy)i)) { + setStrategyMute((routing_strategy)i, true, outputDesc); // do tempMute unmute after twice the mute wait time - setStrategyMute((routing_strategy)i, false, outputDesc->mIoHandle, + setStrategyMute((routing_strategy)i, false, outputDesc, muteWaitMs *2, device); } } @@ -4890,36 +3964,35 @@ uint32_t AudioPolicyManager::checkDeviceMuteStrategies(sp<AudioOutputDescriptor> return 0; } -uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, +uint32_t AudioPolicyManager::setOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, bool force, int delayMs, audio_patch_handle_t *patchHandle, const char* address) { - ALOGV("setOutputDevice() output %d device %04x delayMs %d", output, device, delayMs); - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + ALOGV("setOutputDevice() device %04x delayMs %d", device, delayMs); AudioParameter param; uint32_t muteWaitMs; if (outputDesc->isDuplicated()) { - muteWaitMs = setOutputDevice(outputDesc->mOutput1->mIoHandle, device, force, delayMs); - muteWaitMs += setOutputDevice(outputDesc->mOutput2->mIoHandle, device, force, delayMs); + muteWaitMs = setOutputDevice(outputDesc->subOutput1(), device, force, delayMs); + muteWaitMs += setOutputDevice(outputDesc->subOutput2(), device, force, delayMs); return muteWaitMs; } // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current // output profile if ((device != AUDIO_DEVICE_NONE) && - ((device & outputDesc->mProfile->mSupportedDevices.types()) == 0)) { + ((device & outputDesc->supportedDevices()) == 0)) { return 0; } // filter devices according to output selected - device = (audio_devices_t)(device & outputDesc->mProfile->mSupportedDevices.types()); + device = (audio_devices_t)(device & outputDesc->supportedDevices()); audio_devices_t prevDevice = outputDesc->mDevice; - ALOGV("setOutputDevice() prevDevice %04x", prevDevice); + ALOGV("setOutputDevice() prevDevice 0x%04x", prevDevice); if (device != AUDIO_DEVICE_NONE) { outputDesc->mDevice = device; @@ -4932,10 +4005,10 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, // AND force is not specified // AND the output is connected by a valid audio patch. // Doing this check here allows the caller to call setOutputDevice() without conditions - if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && !force && - outputDesc->mPatchHandle != 0) { - ALOGV("setOutputDevice() setting same device %04x or null device for output %d", - device, output); + if ((device == AUDIO_DEVICE_NONE || device == prevDevice) && + !force && + outputDesc->mPatchHandle != 0) { + ALOGV("setOutputDevice() setting same device 0x%04x or null device", device); return muteWaitMs; } @@ -4943,7 +4016,7 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, // do the routing if (device == AUDIO_DEVICE_NONE) { - resetOutputDevice(output, delayMs, NULL); + resetOutputDevice(outputDesc, delayMs, NULL); } else { DeviceVector deviceList = (address == NULL) ? mAvailableOutputDevices.getDevicesFromType(device) @@ -4978,8 +4051,7 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, status, afPatchHandle, patch.num_sources, patch.num_sinks); if (status == NO_ERROR) { if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); + patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; @@ -4998,7 +4070,7 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, // inform all input as well for (size_t i = 0; i < mInputs.size(); i++) { const sp<AudioInputDescriptor> inputDescriptor = mInputs.valueAt(i); - if (!isVirtualInputDevice(inputDescriptor->mDevice)) { + if (!is_virtual_input_device(inputDescriptor->mDevice)) { AudioParameter inputCmd = AudioParameter(); ALOGV("%s: inform input %d of device:%d", __func__, inputDescriptor->mIoHandle, device); @@ -5011,16 +4083,15 @@ uint32_t AudioPolicyManager::setOutputDevice(audio_io_handle_t output, } // update stream volumes according to new device - applyStreamVolumes(output, device, delayMs); + applyStreamVolumes(outputDesc, device, delayMs); return muteWaitMs; } -status_t AudioPolicyManager::resetOutputDevice(audio_io_handle_t output, +status_t AudioPolicyManager::resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, int delayMs, audio_patch_handle_t *patchHandle) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); ssize_t index; if (patchHandle) { index = mAudioPatches.indexOfKey(*patchHandle); @@ -5085,8 +4156,7 @@ status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input, status, afPatchHandle); if (status == NO_ERROR) { if (index < 0) { - patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(), - &patch, mUidCached); + patchDesc = new AudioPatch(&patch, mUidCached); addAudioPatch(patchDesc->mHandle, patchDesc); } else { patchDesc->mPatch = patch; @@ -5128,12 +4198,12 @@ status_t AudioPolicyManager::resetInputDevice(audio_io_handle_t input, return status; } -sp<AudioPolicyManager::IOProfile> AudioPolicyManager::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> AudioPolicyManager::getInputProfile(audio_devices_t device, + String8 address, + uint32_t& samplingRate, + audio_format_t format, + audio_channel_mask_t channelMask, + audio_input_flags_t flags) { // Choose an input profile based on the requested capture parameters: select the first available // profile supporting all requested parameters. @@ -5160,500 +4230,28 @@ sp<AudioPolicyManager::IOProfile> AudioPolicyManager::getInputProfile(audio_devi audio_devices_t AudioPolicyManager::getDeviceAndMixForInputSource(audio_source_t inputSource, - AudioMix **policyMix) + AudioMix **policyMix) { - audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & - ~AUDIO_DEVICE_BIT_IN; + audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN; + audio_devices_t selectedDeviceFromMix = + mPolicyMixes.getDeviceAndMixForInputSource(inputSource, availableDeviceTypes, policyMix); - for (size_t i = 0; i < mPolicyMixes.size(); i++) { - if (mPolicyMixes[i]->mMix.mMixType != MIX_TYPE_RECORDERS) { - continue; - } - for (size_t j = 0; j < mPolicyMixes[i]->mMix.mCriteria.size(); j++) { - if ((RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource == inputSource) || - (RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET == mPolicyMixes[i]->mMix.mCriteria[j].mRule && - mPolicyMixes[i]->mMix.mCriteria[j].mAttr.mSource != inputSource)) { - if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { - if (policyMix != NULL) { - *policyMix = &mPolicyMixes[i]->mMix; - } - return AUDIO_DEVICE_IN_REMOTE_SUBMIX; - } - break; - } - } + if (selectedDeviceFromMix != AUDIO_DEVICE_NONE) { + return selectedDeviceFromMix; } - return getDeviceForInputSource(inputSource); } audio_devices_t AudioPolicyManager::getDeviceForInputSource(audio_source_t inputSource) { - uint32_t device = AUDIO_DEVICE_NONE; - audio_devices_t availableDeviceTypes = mAvailableInputDevices.types() & - ~AUDIO_DEVICE_BIT_IN; - - switch (inputSource) { - case AUDIO_SOURCE_VOICE_UPLINK: - if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { - device = AUDIO_DEVICE_IN_VOICE_CALL; - break; - } - break; - - case AUDIO_SOURCE_DEFAULT: - case AUDIO_SOURCE_MIC: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) { - device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP; - } else if ((mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO) && - (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - - case AUDIO_SOURCE_VOICE_COMMUNICATION: - // Allow only use of devices on primary input if in call and HAL does not support routing - // to voice call path. - if ((mPhoneState == AUDIO_MODE_IN_CALL) && - (mAvailableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) { - availableDeviceTypes = availablePrimaryInputDevices() & ~AUDIO_DEVICE_BIT_IN; - } - - switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) { - case AUDIO_POLICY_FORCE_BT_SCO: - // if SCO device is requested but no SCO device is available, fall back to default case - if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - break; - } - // FALL THROUGH - - default: // FORCE_NONE - if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - - case AUDIO_POLICY_FORCE_SPEAKER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { - device = AUDIO_DEVICE_IN_BACK_MIC; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - } - break; - - case AUDIO_SOURCE_VOICE_RECOGNITION: - case AUDIO_SOURCE_HOTWORD: - if (mForceUse[AUDIO_POLICY_FORCE_FOR_RECORD] == AUDIO_POLICY_FORCE_BT_SCO && - availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) { - device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) { - device = AUDIO_DEVICE_IN_WIRED_HEADSET; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) { - device = AUDIO_DEVICE_IN_USB_DEVICE; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_CAMCORDER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) { - device = AUDIO_DEVICE_IN_BACK_MIC; - } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) { - device = AUDIO_DEVICE_IN_BUILTIN_MIC; - } - break; - case AUDIO_SOURCE_VOICE_DOWNLINK: - case AUDIO_SOURCE_VOICE_CALL: - if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) { - device = AUDIO_DEVICE_IN_VOICE_CALL; - } - break; - case AUDIO_SOURCE_REMOTE_SUBMIX: - if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) { - device = AUDIO_DEVICE_IN_REMOTE_SUBMIX; - } - break; - case AUDIO_SOURCE_FM_TUNER: - if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) { - device = AUDIO_DEVICE_IN_FM_TUNER; - } - break; - default: - ALOGW("getDeviceForInputSource() invalid input source %d", inputSource); - break; - } - ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device); - return device; -} - -bool AudioPolicyManager::isVirtualInputDevice(audio_devices_t device) -{ - if ((device & AUDIO_DEVICE_BIT_IN) != 0) { - device &= ~AUDIO_DEVICE_BIT_IN; - if ((popcount(device) == 1) && ((device & ~APM_AUDIO_IN_DEVICE_VIRTUAL_ALL) == 0)) - return true; - } - return false; -} - -bool AudioPolicyManager::deviceDistinguishesOnAddress(audio_devices_t device) { - return ((device & APM_AUDIO_DEVICE_MATCH_ADDRESS_ALL & ~AUDIO_DEVICE_BIT_IN) != 0); -} - -audio_io_handle_t AudioPolicyManager::getActiveInput(bool ignoreVirtualInputs) -{ - for (size_t i = 0; i < mInputs.size(); i++) { - const sp<AudioInputDescriptor> input_descriptor = mInputs.valueAt(i); - if ((input_descriptor->mRefCount > 0) - && (!ignoreVirtualInputs || !isVirtualInputDevice(input_descriptor->mDevice))) { - return mInputs.keyAt(i); - } - } - return 0; -} - -uint32_t AudioPolicyManager::activeInputsCount() const -{ - uint32_t count = 0; - for (size_t i = 0; i < mInputs.size(); i++) { - const sp<AudioInputDescriptor> desc = mInputs.valueAt(i); - if (desc->mRefCount > 0) { - count++; - } - } - return count; -} - - -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++) { - mStreams[i].mVolumeCurve[j] = - 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; - } + return mEngine->getDeviceForInputSource(inputSource); } float AudioPolicyManager::computeVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, audio_devices_t device) { - float volume = 1.0; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); - StreamDescriptor &streamDesc = mStreams[stream]; - - if (device == AUDIO_DEVICE_NONE) { - device = outputDesc->device(); - } - - volume = volIndexToAmpl(device, streamDesc, index); + float volumeDb = mEngine->volIndexToDb(Volume::getDeviceCategory(device), stream, index); // if a headset is connected, apply the following rules to ring tones and notifications // to avoid sound level bursts in user's ears: @@ -5669,95 +4267,75 @@ float AudioPolicyManager::computeVolume(audio_stream_type_t stream, || (stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL) || (stream == AUDIO_STREAM_SYSTEM) || ((stream_strategy == STRATEGY_ENFORCED_AUDIBLE) && - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) && - streamDesc.mCanBeMuted) { - volume *= SONIFICATION_HEADSET_VOLUME_FACTOR; + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) && + mStreams.canBeMuted(stream)) { + volumeDb += SONIFICATION_HEADSET_VOLUME_FACTOR_DB; // when the phone is ringing we must consider that music could have been paused just before // by the music application and behave as if music was active if the last music track was // just stopped if (isStreamActive(AUDIO_STREAM_MUSIC, SONIFICATION_HEADSET_MUSIC_DELAY) || mLimitRingtoneVolume) { audio_devices_t musicDevice = getDeviceForStrategy(STRATEGY_MEDIA, true /*fromCache*/); - float musicVol = computeVolume(AUDIO_STREAM_MUSIC, - mStreams[AUDIO_STREAM_MUSIC].getVolumeIndex(musicDevice), - output, + float musicVolDB = computeVolume(AUDIO_STREAM_MUSIC, + mStreams.valueFor(AUDIO_STREAM_MUSIC).getVolumeIndex(musicDevice), musicDevice); - float minVol = (musicVol > SONIFICATION_HEADSET_VOLUME_MIN) ? - musicVol : SONIFICATION_HEADSET_VOLUME_MIN; - if (volume > minVol) { - volume = minVol; - ALOGV("computeVolume limiting volume to %f musicVol %f", minVol, musicVol); + float minVolDB = (musicVolDB > SONIFICATION_HEADSET_VOLUME_MIN_DB) ? + musicVolDB : SONIFICATION_HEADSET_VOLUME_MIN_DB; + if (volumeDb > minVolDB) { + volumeDb = minVolDB; + ALOGV("computeVolume limiting volume to %f musicVol %f", minVolDB, musicVolDB); } } } - return volume; + return volumeDb; } status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, int index, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, int delayMs, bool force) { - // do not change actual stream volume if the stream is muted - if (mOutputs.valueFor(output)->mMuteCount[stream] != 0) { + if (outputDesc->mMuteCount[stream] != 0) { ALOGVV("checkAndSetVolume() stream %d muted count %d", - stream, mOutputs.valueFor(output)->mMuteCount[stream]); + stream, outputDesc->mMuteCount[stream]); return NO_ERROR; } - + audio_policy_forced_cfg_t forceUseForComm = + mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION); // do not change in call volume if bluetooth is connected and vice versa - if ((stream == AUDIO_STREAM_VOICE_CALL && - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] == AUDIO_POLICY_FORCE_BT_SCO) || - (stream == AUDIO_STREAM_BLUETOOTH_SCO && - mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION] != AUDIO_POLICY_FORCE_BT_SCO)) { + if ((stream == AUDIO_STREAM_VOICE_CALL && forceUseForComm == AUDIO_POLICY_FORCE_BT_SCO) || + (stream == AUDIO_STREAM_BLUETOOTH_SCO && forceUseForComm != AUDIO_POLICY_FORCE_BT_SCO)) { ALOGV("checkAndSetVolume() cannot set stream %d volume with force use = %d for comm", - stream, mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]); + stream, forceUseForComm); return INVALID_OPERATION; } - float volume = computeVolume(stream, index, output, device); - // unit gain if rerouting to external policy - if (device == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { - ssize_t index = mOutputs.indexOfKey(output); - if (index >= 0) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueAt(index); - if (outputDesc->mPolicyMix != NULL) { - ALOGV("max gain when rerouting for output=%d", output); - volume = 1.0f; - } - } - + if (device == AUDIO_DEVICE_NONE) { + device = outputDesc->device(); } - // We actually change the volume if: - // - the float value returned by computeVolume() changed - // - the force flag is set - if (volume != mOutputs.valueFor(output)->mCurVolume[stream] || - force) { - mOutputs.valueFor(output)->mCurVolume[stream] = volume; - ALOGVV("checkAndSetVolume() for output %d stream %d, volume %f, delay %d", output, stream, volume, delayMs); - // Force VOICE_CALL to track BLUETOOTH_SCO stream volume when bluetooth audio is - // enabled - if (stream == AUDIO_STREAM_BLUETOOTH_SCO) { - mpClientInterface->setStreamVolume(AUDIO_STREAM_VOICE_CALL, volume, output, delayMs); - } - mpClientInterface->setStreamVolume(stream, volume, output, delayMs); + + float volumeDb = computeVolume(stream, index, device); + if (outputDesc->isFixedVolume(device)) { + volumeDb = 0.0f; } + outputDesc->setVolume(volumeDb, stream, device, delayMs, force); + if (stream == AUDIO_STREAM_VOICE_CALL || stream == AUDIO_STREAM_BLUETOOTH_SCO) { float voiceVolume; // Force voice volume to max for bluetooth SCO as volume is managed by the headset if (stream == AUDIO_STREAM_VOICE_CALL) { - voiceVolume = (float)index/(float)mStreams[stream].mIndexMax; + voiceVolume = (float)index/(float)mStreams.valueFor(stream).getVolumeIndexMax(); } else { voiceVolume = 1.0; } - if (voiceVolume != mLastVoiceVolume && output == mPrimaryOutput) { + if (voiceVolume != mLastVoiceVolume && outputDesc == mPrimaryOutput) { mpClientInterface->setVoiceVolume(voiceVolume, delayMs); mLastVoiceVolume = voiceVolume; } @@ -5766,20 +4344,20 @@ status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream, return NO_ERROR; } -void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, +void AudioPolicyManager::applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc, audio_devices_t device, int delayMs, bool force) { - ALOGVV("applyStreamVolumes() for output %d and device %x", output, device); + ALOGVV("applyStreamVolumes() for device %08x", device); for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { if (stream == AUDIO_STREAM_PATCH) { continue; } checkAndSetVolume((audio_stream_type_t)stream, - mStreams[stream].getVolumeIndex(device), - output, + mStreams.valueFor((audio_stream_type_t)stream).getVolumeIndex(device), + outputDesc, device, delayMs, force); @@ -5788,7 +4366,7 @@ void AudioPolicyManager::applyStreamVolumes(audio_io_handle_t output, void AudioPolicyManager::setStrategyMute(routing_strategy strategy, bool on, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, int delayMs, audio_devices_t device) { @@ -5798,32 +4376,31 @@ void AudioPolicyManager::setStrategyMute(routing_strategy strategy, continue; } if (getStrategy((audio_stream_type_t)stream) == strategy) { - setStreamMute((audio_stream_type_t)stream, on, output, delayMs, device); + setStreamMute((audio_stream_type_t)stream, on, outputDesc, delayMs, device); } } } void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, bool on, - audio_io_handle_t output, + const sp<AudioOutputDescriptor>& outputDesc, int delayMs, audio_devices_t device) { - StreamDescriptor &streamDesc = mStreams[stream]; - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output); + const StreamDescriptor& streamDesc = mStreams.valueFor(stream); if (device == AUDIO_DEVICE_NONE) { device = outputDesc->device(); } - ALOGVV("setStreamMute() stream %d, mute %d, output %d, mMuteCount %d device %04x", - stream, on, output, outputDesc->mMuteCount[stream], device); + ALOGVV("setStreamMute() stream %d, mute %d, mMuteCount %d device %04x", + stream, on, outputDesc->mMuteCount[stream], device); if (on) { if (outputDesc->mMuteCount[stream] == 0) { - if (streamDesc.mCanBeMuted && + if (streamDesc.canBeMuted() && ((stream != AUDIO_STREAM_ENFORCED_AUDIBLE) || - (mForceUse[AUDIO_POLICY_FORCE_FOR_SYSTEM] == AUDIO_POLICY_FORCE_NONE))) { - checkAndSetVolume(stream, 0, output, device, delayMs); + (mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_NONE))) { + checkAndSetVolume(stream, 0, outputDesc, device, delayMs); } } // increment mMuteCount after calling checkAndSetVolume() so that volume change is not ignored @@ -5836,7 +4413,7 @@ void AudioPolicyManager::setStreamMute(audio_stream_type_t stream, if (--outputDesc->mMuteCount[stream] == 0) { checkAndSetVolume(stream, streamDesc.getVolumeIndex(device), - output, + outputDesc, device, delayMs); } @@ -5855,7 +4432,7 @@ void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, const routing_strategy stream_strategy = getStrategy(stream); if ((stream_strategy == STRATEGY_SONIFICATION) || ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) { - sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(mPrimaryOutput); + sp<SwAudioOutputDescriptor> outputDesc = mPrimaryOutput; ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d", stream, starting, outputDesc->mDevice, stateChange); if (outputDesc->mRefCount[stream]) { @@ -5888,2109 +4465,83 @@ void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream, } } -bool AudioPolicyManager::isInCall() -{ - return isStateInCall(mPhoneState); +// --- SessionRoute class implementation +void AudioPolicyManager::SessionRoute::log(const char* prefix) { + ALOGI("%s[SessionRoute strm:0x%X, sess:0x%X, dev:0x%X refs:%d act:%d", + prefix, mStreamType, mSession, + mDeviceDescriptor != 0 ? mDeviceDescriptor->type() : AUDIO_DEVICE_NONE, + mRefCount, mActivityCount); } -bool AudioPolicyManager::isStateInCall(int state) { - return ((state == AUDIO_MODE_IN_CALL) || - (state == AUDIO_MODE_IN_COMMUNICATION)); -} - -uint32_t AudioPolicyManager::getMaxEffectsCpuLoad() +// --- SessionRouteMap class implementation +bool AudioPolicyManager::SessionRouteMap::hasRoute(audio_session_t session) { - return MAX_EFFECTS_CPU_LOAD; + return indexOfKey(session) >= 0 && valueFor(session)->mDeviceDescriptor != 0; } -uint32_t AudioPolicyManager::getMaxEffectsMemory() +void AudioPolicyManager::SessionRouteMap::addRoute(audio_session_t session, + audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor) { - return MAX_EFFECTS_MEMORY; -} - - -// --- 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()); + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != NULL) { + route->mRefCount++; + route->mDeviceDescriptor = deviceDescriptor; } else { - return mProfile->mSupportedDevices.types() ; + route = new AudioPolicyManager::SessionRoute(session, streamType, deviceDescriptor); + route->mRefCount++; + add(session, route); } } -bool AudioPolicyManager::AudioOutputDescriptor::isActive(uint32_t inPastMs) const +void AudioPolicyManager::SessionRouteMap::removeRoute(audio_session_t session) { - 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; + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != 0) { + ALOG_ASSERT(route->mRefCount > 0); + --route->mRefCount; + if (route->mRefCount <= 0) { + removeItem(session); } } - return false; } -bool AudioPolicyManager::AudioOutputDescriptor::isStreamActive(audio_stream_type_t stream, - uint32_t inPastMs, - nsecs_t sysTime) const +int AudioPolicyManager::SessionRouteMap::incRouteActivity(audio_session_t session) { - 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()); + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + return route != 0 ? ++(route->mActivityCount) : -1; } -// --- EffectDescriptor class implementation - -status_t AudioPolicyManager::EffectDescriptor::dump(int fd) +int AudioPolicyManager::SessionRouteMap::decRouteActivity(audio_session_t session) { - const size_t SIZE = 256; - char buffer[SIZE]; - String8 result; - - snprintf(buffer, SIZE, " I/O: %d\n", mIo); - result.append(buffer); - snprintf(buffer, SIZE, " Strategy: %d\n", mStrategy); - result.append(buffer); - snprintf(buffer, SIZE, " Session: %d\n", mSession); - result.append(buffer); - snprintf(buffer, SIZE, " Name: %s\n", mDesc.name); - result.append(buffer); - snprintf(buffer, SIZE, " %s\n", mEnabled ? "Enabled" : "Disabled"); - result.append(buffer); - write(fd, result.string(), result.size()); - - 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; + sp<SessionRoute> route = indexOfKey(session) >= 0 ? valueFor(session) : 0; + if (route != 0 && route->mActivityCount > 0) { + return --(route->mActivityCount); } 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(String8(""), 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(String8(""), 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) -{ - mUseInChannelMask = ((type == AUDIO_PORT_TYPE_DEVICE) && (role == AUDIO_PORT_ROLE_SOURCE)) || - ((type == AUDIO_PORT_TYPE_MIX) && (role == AUDIO_PORT_ROLE_SINK)); -} - -void AudioPolicyManager::AudioPort::toAudioPort(struct audio_port *port) const -{ - port->role = mRole; - port->type = mType; - 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; +void AudioPolicyManager::SessionRouteMap::log(const char* caption) { + ALOGI("%s ----", caption); + for(size_t index = 0; index < size(); index++) { + valueAt(index)->log(" "); } } -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 - - -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(""), mId(0) -{ -} - -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(""), 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(""), 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++) { - ALOGV("DeviceVector::getDeviceFromId(%d) itemAt(%zu)->mId %d", id, i, itemAt(i)->mId); - if (itemAt(i)->mId == id) { - device = itemAt(i); - break; - } - } - return device; -} - -AudioPolicyManager::DeviceVector AudioPolicyManager::DeviceVector::getDevicesFromType( - audio_devices_t type) const -{ - DeviceVector devices; - bool isOutput = audio_is_output_devices(type); - type &= ~AUDIO_DEVICE_BIT_IN; - for (size_t i = 0; (i < size()) && (type != AUDIO_DEVICE_NONE); i++) { - bool curIsOutput = audio_is_output_devices(itemAt(i)->mDeviceType); - audio_devices_t curType = itemAt(i)->mDeviceType & ~AUDIO_DEVICE_BIT_IN; - if ((isOutput == curIsOutput) && ((type & curType) != 0)) { - devices.add(itemAt(i)); - type &= ~curType; - 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; - dstConfig->ext.device.hw_module = mModule->mHandle; - 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; -} - -void AudioPolicyManager::loadHwModule(cnode *root) -{ - status_t status = NAME_NOT_FOUND; - cnode *node; - sp<HwModule> module = new HwModule(root->name); - - node = config_find(root, DEVICES_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading device %s", node->name); - status_t tmpStatus = module->loadDevice(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - node = config_find(root, OUTPUTS_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading output %s", node->name); - status_t tmpStatus = module->loadOutput(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - node = config_find(root, INPUTS_TAG); - if (node != NULL) { - node = node->first_child; - while (node) { - ALOGV("loadHwModule() loading input %s", node->name); - status_t tmpStatus = module->loadInput(node); - if (status == NAME_NOT_FOUND || status == NO_ERROR) { - status = tmpStatus; - } - node = node->next; - } - } - loadGlobalConfig(root, module); - - if (status == NO_ERROR) { - mHwModules.add(module); - } -} - -void AudioPolicyManager::loadHwModules(cnode *root) -{ - cnode *node = config_find(root, AUDIO_HW_MODULE_TAG); - if (node == NULL) { - return; - } - - node = node->first_child; - while (node) { - ALOGV("loadHwModules() loading module %s", node->name); - loadHwModule(node); - node = node->next; - } -} - -void AudioPolicyManager::loadGlobalConfig(cnode *root, const sp<HwModule>& module) -{ - cnode *node = config_find(root, GLOBAL_CONFIG_TAG); - - if (node == NULL) { - return; - } - DeviceVector declaredDevices; - if (module != NULL) { - declaredDevices = module->mDeclaredDevices; - } - - node = node->first_child; - while (node) { - if (strcmp(ATTACHED_OUTPUT_DEVICES_TAG, node->name) == 0) { - mAvailableOutputDevices.loadDevicesFromName((char *)node->value, - declaredDevices); - 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); - if (device != AUDIO_DEVICE_NONE) { - mDefaultOutputDevice = new DeviceDescriptor(String8(""), device); - } else { - ALOGW("loadGlobalConfig() default device not specified"); - } - ALOGV("loadGlobalConfig() mDefaultOutputDevice %08x", mDefaultOutputDevice->mDeviceType); - } else if (strcmp(ATTACHED_INPUT_DEVICES_TAG, node->name) == 0) { - mAvailableInputDevices.loadDevicesFromName((char *)node->value, - declaredDevices); - ALOGV("loadGlobalConfig() Available InputDevices %08x", mAvailableInputDevices.types()); - } else if (strcmp(SPEAKER_DRC_ENABLED_TAG, node->name) == 0) { - mSpeakerDrcEnabled = stringToBool((char *)node->value); - ALOGV("loadGlobalConfig() mSpeakerDrcEnabled = %d", mSpeakerDrcEnabled); - } else if (strcmp(AUDIO_HAL_VERSION_TAG, node->name) == 0) { - uint32_t major, minor; - sscanf((char *)node->value, "%u.%u", &major, &minor); - module->mHalVersion = HARDWARE_DEVICE_API_VERSION(major, minor); - ALOGV("loadGlobalConfig() mHalVersion = %04x major %u minor %u", - module->mHalVersion, major, minor); - } - node = node->next; - } -} - -status_t AudioPolicyManager::loadAudioPolicyConfig(const char *path) -{ - cnode *root; - char *data; - - data = (char *)load_file(path, NULL); - if (data == NULL) { - return -ENODEV; - } - root = config_node("", ""); - config_load(root, data); - - loadHwModules(root); - // legacy audio_policy.conf files have one global_configuration section - loadGlobalConfig(root, getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)); - config_free(root); - free(root); - free(data); - - ALOGI("loadAudioPolicyConfig() loaded %s\n", path); - - return NO_ERROR; -} - void AudioPolicyManager::defaultAudioPolicyConfig(void) { sp<HwModule> module; sp<IOProfile> profile; - sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(String8(""), - AUDIO_DEVICE_IN_BUILTIN_MIC); + sp<DeviceDescriptor> defaultInputDevice = + new DeviceDescriptor(String8("builtin-mic"), AUDIO_DEVICE_IN_BUILTIN_MIC); mAvailableOutputDevices.add(mDefaultOutputDevice); mAvailableInputDevices.add(defaultInputDevice); module = new HwModule("primary"); - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE, module); + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SOURCE); + profile->attach(module); profile->mSamplingRates.add(44100); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_OUT_STEREO); @@ -7998,7 +4549,8 @@ void AudioPolicyManager::defaultAudioPolicyConfig(void) profile->mFlags = AUDIO_OUTPUT_FLAG_PRIMARY; module->mOutputProfiles.add(profile); - profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK, module); + profile = new IOProfile(String8("primary"), AUDIO_PORT_ROLE_SINK); + profile->attach(module); profile->mSamplingRates.add(8000); profile->mFormats.add(AUDIO_FORMAT_PCM_16_BIT); profile->mChannelMasks.add(AUDIO_CHANNEL_IN_MONO); @@ -8064,7 +4616,8 @@ audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_ } } -bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { +bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) +{ // has flags that map to a strategy? if ((paa->flags & (AUDIO_FLAG_AUDIBILITY_ENFORCED | AUDIO_FLAG_SCO | AUDIO_FLAG_BEACON)) != 0) { return true; @@ -8095,4 +4648,39 @@ bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa) { return true; } +bool AudioPolicyManager::isStrategyActive(const sp<AudioOutputDescriptor> outputDesc, + 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)) && + outputDesc->isStreamActive((audio_stream_type_t)i, inPastMs, sysTime)) { + return true; + } + } + return false; +} + +audio_policy_forced_cfg_t AudioPolicyManager::getForceUse(audio_policy_force_use_t usage) +{ + return mEngine->getForceUse(usage); +} + +bool AudioPolicyManager::isInCall() +{ + return isStateInCall(mEngine->getPhoneState()); +} + +bool AudioPolicyManager::isStateInCall(int state) +{ + return is_state_in_call(state); +} + }; // namespace android diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h new file mode 100644 index 0000000..11fd5ff --- /dev/null +++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h @@ -0,0 +1,632 @@ +/* + * 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. + */ + +#pragma once + +#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 <AudioPolicyManagerInterface.h> +#include <AudioPolicyManagerObserver.h> +#include <AudioGain.h> +#include <AudioPort.h> +#include <AudioPatch.h> +#include <ConfigParsingUtils.h> +#include <DeviceDescriptor.h> +#include <IOProfile.h> +#include <HwModule.h> +#include <AudioInputDescriptor.h> +#include <AudioOutputDescriptor.h> +#include <AudioPolicyMix.h> +#include <EffectDescriptor.h> +#include <SoundTriggerSession.h> +#include <StreamDescriptor.h> + +namespace android { + +// ---------------------------------------------------------------------------- + +// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB +#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5 +#define SONIFICATION_HEADSET_VOLUME_FACTOR_DB (-6) +// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB +#define SONIFICATION_HEADSET_VOLUME_MIN 0.016 +#define SONIFICATION_HEADSET_VOLUME_MIN_DB (-36) + +// 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 during witch some streams are muted while the audio path +// is switched +#define MUTE_TIME_MS 2000 + +#define NUM_TEST_OUTPUTS 5 + +#define NUM_VOL_CURVE_KNEES 2 + +// Default minimum length allowed for offloading a compressed track +// Can be overridden by the audio.offload.min.duration.secs property +#define OFFLOAD_DEFAULT_MIN_DURATION_SECS 60 + +// ---------------------------------------------------------------------------- +// AudioPolicyManager implements audio policy manager behavior common to all platforms. +// ---------------------------------------------------------------------------- + +class AudioPolicyManager : public AudioPolicyInterface, public AudioPolicyManagerObserver + +#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, + audio_port_handle_t selectedDeviceId, + 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) + { + return mEffects.unregisterEffect(id); + } + virtual status_t setEffectEnabled(int id, bool enabled) + { + return mEffects.setEffectEnabled(id, 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) + { + return mSoundTriggerSessions.releaseSession(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 defaultAudioPolicyConfig(void); + + // return the strategy corresponding to a given stream type + routing_strategy getStrategy(audio_stream_type_t stream) const; + +protected: + class SessionRoute : public RefBase + { + public: + friend class SessionRouteMap; + SessionRoute(audio_session_t session, + audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor) + : mSession(session), + mStreamType(streamType), + mDeviceDescriptor(deviceDescriptor), + mRefCount(0), + mActivityCount(0) {} + + audio_session_t mSession; + audio_stream_type_t mStreamType; + + sp<DeviceDescriptor> mDeviceDescriptor; + + // "reference" counting + int mRefCount; // +/- on references + int mActivityCount; // +/- on start/stop + + void log(const char* prefix); + }; + + class SessionRouteMap: public KeyedVector<audio_session_t, sp<SessionRoute>> + { + public: + bool hasRoute(audio_session_t session); + void addRoute(audio_session_t session, audio_stream_type_t streamType, + sp<DeviceDescriptor> deviceDescriptor); + void removeRoute(audio_session_t session); + + int incRouteActivity(audio_session_t session); + int decRouteActivity(audio_session_t session); + + void log(const char* caption); + }; + + // From AudioPolicyManagerObserver + virtual const AudioPatchCollection &getAudioPatches() const + { + return mAudioPatches; + } + virtual const SoundTriggerSessionCollection &getSoundTriggerSessionCollection() const + { + return mSoundTriggerSessions; + } + virtual const AudioPolicyMixCollection &getAudioPolicyMixCollection() const + { + return mPolicyMixes; + } + virtual const SwAudioOutputCollection &getOutputs() const + { + return mOutputs; + } + virtual const AudioInputCollection &getInputs() const + { + return mInputs; + } + virtual const DeviceVector &getAvailableOutputDevices() const + { + return mAvailableOutputDevices; + } + virtual const DeviceVector &getAvailableInputDevices() const + { + return mAvailableInputDevices; + } + virtual StreamDescriptorCollection &getStreamDescriptors() + { + return mStreams; + } + virtual const sp<DeviceDescriptor> &getDefaultOutputDevice() const + { + return mDefaultOutputDevice; + } +protected: + void addOutput(audio_io_handle_t output, sp<SwAudioOutputDescriptor> outputDesc); + void removeOutput(audio_io_handle_t output); + 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); + + bool isStrategyActive(const sp<AudioOutputDescriptor> outputDesc, routing_strategy strategy, + uint32_t inPastMs = 0, nsecs_t sysTime = 0) const; + + // 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(const sp<AudioOutputDescriptor>& outputDesc, + audio_devices_t device, + bool force = false, + int delayMs = 0, + audio_patch_handle_t *patchHandle = NULL, + const char* address = NULL); + status_t resetOutputDevice(const sp<AudioOutputDescriptor>& outputDesc, + 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); + + // 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_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, + const sp<AudioOutputDescriptor>& outputDesc, + audio_devices_t device, + int delayMs = 0, bool force = false); + + // apply all stream volumes to the specified output and device + void applyStreamVolumes(const sp<AudioOutputDescriptor>& outputDesc, + 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, + const sp<AudioOutputDescriptor>& outputDesc, + 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, + const sp<AudioOutputDescriptor>& outputDesc, + 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); + + audio_mode_t getPhoneState(); + + // true if device is in a telephony or VoIP call + virtual bool isInCall(); + // true if given state represents a device in a telephony or VoIP call + virtual bool isStateInCall(int state); + + // 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(); + + // 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(const sp<AudioOutputDescriptor>& outputDesc, + 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() + { + return mEffects.getMaxEffectsCpuLoad(); + } + + virtual uint32_t getMaxEffectsMemory() + { + return mEffects.getMaxEffectsMemory(); + } +#ifdef AUDIO_POLICY_TEST + virtual bool threadLoop(); + void exit(); + int testOutputIndex(audio_io_handle_t output); +#endif //AUDIO_POLICY_TEST + + SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device, + SwAudioOutputCollection 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); + + virtual status_t addAudioPatch(audio_patch_handle_t handle, const sp<AudioPatch>& patch) + { + return mAudioPatches.addAudioPatch(handle, patch); + } + virtual status_t removeAudioPatch(audio_patch_handle_t handle) + { + return mAudioPatches.removeAudioPatch(handle); + } + + audio_devices_t availablePrimaryOutputDevices() const + { + return mPrimaryOutput->supportedDevices() & mAvailableOutputDevices.types(); + } + audio_devices_t availablePrimaryInputDevices() const + { + return mAvailableInputDevices.getDevicesFromHwModule(mPrimaryOutput->getModuleHandle()); + } + + void updateCallRouting(audio_devices_t rxDevice, int delayMs = 0); + + status_t startSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream, + audio_devices_t device, + uint32_t *delayMs); + status_t stopSource(sp<AudioOutputDescriptor> outputDesc, + audio_stream_type_t stream); + + uid_t mUidCached; + AudioPolicyClientInterface *mpClientInterface; // audio policy client interface + sp<SwAudioOutputDescriptor> mPrimaryOutput; // primary output descriptor + // list of descriptors for outputs currently opened + + SwAudioOutputCollection mOutputs; + // copy of mOutputs before setDeviceConnectionState() opens new outputs + // reset to mOutputs when updateDevicesAndOutputs() is called. + SwAudioOutputCollection mPreviousOutputs; + AudioInputCollection mInputs; // list of input descriptors + + DeviceVector mAvailableOutputDevices; // all available output devices + DeviceVector mAvailableInputDevices; // all available input devices + + SessionRouteMap mOutputRoutes; + SessionRouteMap mInputRoutes; + + StreamDescriptorCollection mStreams; // 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 + + EffectDescriptorCollection 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 + + HwModuleCollection mHwModules; + + volatile int32_t mAudioPortGeneration; + + AudioPatchCollection mAudioPatches; + + SoundTriggerSessionCollection 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 + + AudioPolicyMixCollection 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 + + uint32_t nextAudioPortGeneration(); + + // Audio Policy Engine Interface. + AudioPolicyManagerInterface *mEngine; +private: + // updates device caching and output for streams that can influence the + // routing of notifications + void handleNotificationRoutingForStream(audio_stream_type_t stream); + // 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<SwAudioOutputDescriptor> 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); + // 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); +}; + +}; diff --git a/services/audiopolicy/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp index 3e090e9..3e090e9 100644 --- a/services/audiopolicy/AudioPolicyClientImpl.cpp +++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp diff --git a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp index a79f8ae..a79f8ae 100644 --- a/services/audiopolicy/AudioPolicyClientImplLegacy.cpp +++ b/services/audiopolicy/service/AudioPolicyClientImplLegacy.cpp diff --git a/services/audiopolicy/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp index e6ace20..e6ace20 100644 --- a/services/audiopolicy/AudioPolicyEffects.cpp +++ b/services/audiopolicy/service/AudioPolicyEffects.cpp diff --git a/services/audiopolicy/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h index 3dec437..3dec437 100644 --- a/services/audiopolicy/AudioPolicyEffects.h +++ b/services/audiopolicy/service/AudioPolicyEffects.h diff --git a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp index a45dbb3..a763151 100644 --- a/services/audiopolicy/AudioPolicyInterfaceImpl.cpp +++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp @@ -28,7 +28,8 @@ namespace android { status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address) + const char *device_address, + const char *device_name) { if (mAudioPolicyManager == NULL) { return NO_INIT; @@ -46,8 +47,8 @@ status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, ALOGV("setDeviceConnectionState()"); Mutex::Autolock _l(mLock); - return mAudioPolicyManager->setDeviceConnectionState(device, - state, device_address); + return mAudioPolicyManager->setDeviceConnectionState(device, state, + device_address, device_name); } audio_policy_dev_state_t AudioPolicyService::getDeviceConnectionState( @@ -149,6 +150,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int mSelectedDeviceId, const audio_offload_info_t *offloadInfo) { if (mAudioPolicyManager == NULL) { @@ -157,7 +159,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, ALOGV("getOutput()"); Mutex::Autolock _l(mLock); return mAudioPolicyManager->getOutputForAttr(attr, output, session, stream, samplingRate, - format, channelMask, flags, offloadInfo); + format, channelMask, flags, mSelectedDeviceId, offloadInfo); } status_t AudioPolicyService::startOutput(audio_io_handle_t output, diff --git a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp index b8846c6..372a9fa 100644 --- a/services/audiopolicy/AudioPolicyInterfaceImplLegacy.cpp +++ b/services/audiopolicy/service/AudioPolicyInterfaceImplLegacy.cpp @@ -33,7 +33,8 @@ namespace android { status_t AudioPolicyService::setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address) + const char *device_address, + const char *device_name __unused) { if (mpAudioPolicy == NULL) { return NO_INIT; @@ -568,6 +569,7 @@ status_t AudioPolicyService::getOutputForAttr(const audio_attributes_t *attr, audio_format_t format, audio_channel_mask_t channelMask, audio_output_flags_t flags, + int selectedDeviceId __unused, const audio_offload_info_t *offloadInfo) { if (attr != NULL) { diff --git a/services/audiopolicy/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp index eb9116d..00f188f 100644 --- a/services/audiopolicy/AudioPolicyService.cpp +++ b/services/audiopolicy/service/AudioPolicyService.cpp @@ -532,7 +532,7 @@ bool AudioPolicyService::AudioCommandThread::threadLoop() mLock.unlock(); svc.clear(); mLock.lock(); - if (!exitPending() && mAudioCommands.isEmpty()) { + if (!exitPending() && (mAudioCommands.isEmpty() || waitTime != INT64_MAX)) { // release delayed commands wake lock release_wake_lock(mName.string()); ALOGV("AudioCommandThread() going to sleep"); diff --git a/services/audiopolicy/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h index 80284a4..f8dabd3 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 { @@ -61,7 +61,8 @@ public: virtual status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, - const char *device_address); + const char *device_address, + const char *device_name); virtual audio_policy_dev_state_t getDeviceConnectionState( audio_devices_t device, const char *device_address); @@ -83,6 +84,7 @@ public: audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = 0, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, + int selectedDeviceId = AUDIO_PORT_HANDLE_NONE, const audio_offload_info_t *offloadInfo = NULL); virtual status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, |