| /* |
| * 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. |
| */ |
| |
| #ifndef ANDROID_AUDIOPOLICYSERVICE_H |
| #define ANDROID_AUDIOPOLICYSERVICE_H |
| |
| #include <cutils/misc.h> |
| #include <cutils/config_utils.h> |
| #include <utils/String8.h> |
| #include <utils/Vector.h> |
| #include <utils/SortedVector.h> |
| #include <binder/BinderService.h> |
| #include <system/audio.h> |
| #include <system/audio_policy.h> |
| #include <hardware/audio_policy.h> |
| #include <media/IAudioPolicyService.h> |
| #include <media/ToneGenerator.h> |
| #include <media/AudioEffect.h> |
| |
| namespace android { |
| |
| // ---------------------------------------------------------------------------- |
| |
| class AudioPolicyService : |
| public BinderService<AudioPolicyService>, |
| public BnAudioPolicyService, |
| // public AudioPolicyClientInterface, |
| public IBinder::DeathRecipient |
| { |
| friend class BinderService<AudioPolicyService>; |
| |
| public: |
| // for BinderService |
| static const char *getServiceName() { return "media.audio_policy"; } |
| |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| // |
| // BnAudioPolicyService (see AudioPolicyInterface for method descriptions) |
| // |
| |
| 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 status_t setPhoneState(audio_mode_t state); |
| virtual status_t 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 audio_io_handle_t getOutput(audio_stream_type_t stream, |
| uint32_t samplingRate = 0, |
| audio_format_t format = AUDIO_FORMAT_DEFAULT, |
| audio_channel_mask_t channelMask = 0, |
| audio_output_flags_t flags = |
| AUDIO_OUTPUT_FLAG_NONE); |
| virtual status_t startOutput(audio_io_handle_t output, |
| audio_stream_type_t stream, |
| int session = 0); |
| virtual status_t stopOutput(audio_io_handle_t output, |
| audio_stream_type_t stream, |
| int session = 0); |
| virtual void releaseOutput(audio_io_handle_t output); |
| virtual audio_io_handle_t getInput(audio_source_t inputSource, |
| uint32_t samplingRate = 0, |
| audio_format_t format = AUDIO_FORMAT_DEFAULT, |
| audio_channel_mask_t channelMask = 0, |
| int audioSession = 0); |
| virtual status_t startInput(audio_io_handle_t input); |
| virtual status_t stopInput(audio_io_handle_t input); |
| virtual void releaseInput(audio_io_handle_t input); |
| virtual status_t 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); |
| |
| virtual uint32_t getStrategyForStream(audio_stream_type_t stream); |
| virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream); |
| |
| virtual audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); |
| 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; |
| virtual bool isStreamActiveRemotely(audio_stream_type_t stream, uint32_t inPastMs = 0) const; |
| virtual bool isSourceActive(audio_source_t source) const; |
| |
| virtual status_t queryDefaultPreProcessing(int audioSession, |
| effect_descriptor_t *descriptors, |
| uint32_t *count); |
| virtual status_t onTransact( |
| uint32_t code, |
| const Parcel& data, |
| Parcel* reply, |
| uint32_t flags); |
| |
| // IBinder::DeathRecipient |
| virtual void binderDied(const wp<IBinder>& who); |
| |
| // |
| // Helpers for the struct audio_policy_service_ops implementation. |
| // This is used by the audio policy manager for certain operations that |
| // are implemented by the policy service. |
| // |
| virtual void setParameters(audio_io_handle_t ioHandle, |
| const char *keyValuePairs, |
| int delayMs); |
| |
| virtual status_t setStreamVolume(audio_stream_type_t stream, |
| float volume, |
| audio_io_handle_t output, |
| int delayMs = 0); |
| virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream); |
| virtual status_t stopTone(); |
| virtual status_t setVoiceVolume(float volume, int delayMs = 0); |
| |
| private: |
| AudioPolicyService(); |
| virtual ~AudioPolicyService(); |
| |
| status_t dumpInternals(int fd); |
| |
| // Thread used for tone playback and to send audio config commands to audio flinger |
| // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because |
| // startTone() and stopTone() are normally called with mLock locked and requesting a tone start |
| // or stop will cause calls to AudioPolicyService and an attempt to lock mLock. |
| // For audio config commands, it is necessary because audio flinger requires that the calling |
| // process (user) has permission to modify audio settings. |
| class AudioCommandThread : public Thread { |
| class AudioCommand; |
| public: |
| |
| // commands for tone AudioCommand |
| enum { |
| START_TONE, |
| STOP_TONE, |
| SET_VOLUME, |
| SET_PARAMETERS, |
| SET_VOICE_VOLUME |
| }; |
| |
| AudioCommandThread (String8 name); |
| virtual ~AudioCommandThread(); |
| |
| status_t dump(int fd); |
| |
| // Thread virtuals |
| virtual void onFirstRef(); |
| virtual bool threadLoop(); |
| |
| void exit(); |
| void startToneCommand(ToneGenerator::tone_type type, |
| audio_stream_type_t stream); |
| void stopToneCommand(); |
| status_t volumeCommand(audio_stream_type_t stream, float volume, |
| audio_io_handle_t output, int delayMs = 0); |
| status_t parametersCommand(audio_io_handle_t ioHandle, |
| const char *keyValuePairs, int delayMs = 0); |
| status_t voiceVolumeCommand(float volume, int delayMs = 0); |
| void insertCommand_l(AudioCommand *command, int delayMs = 0); |
| |
| private: |
| // descriptor for requested tone playback event |
| class AudioCommand { |
| |
| public: |
| AudioCommand() |
| : mCommand(-1) {} |
| |
| void dump(char* buffer, size_t size); |
| |
| int mCommand; // START_TONE, STOP_TONE ... |
| nsecs_t mTime; // time stamp |
| Condition mCond; // condition for status return |
| status_t mStatus; // command status |
| bool mWaitStatus; // true if caller is waiting for status |
| void *mParam; // command parameter (ToneData, VolumeData, ParametersData) |
| }; |
| |
| class ToneData { |
| public: |
| ToneGenerator::tone_type mType; // tone type (START_TONE only) |
| audio_stream_type_t mStream; // stream type (START_TONE only) |
| }; |
| |
| class VolumeData { |
| public: |
| audio_stream_type_t mStream; |
| float mVolume; |
| audio_io_handle_t mIO; |
| }; |
| |
| class ParametersData { |
| public: |
| audio_io_handle_t mIO; |
| String8 mKeyValuePairs; |
| }; |
| |
| class VoiceVolumeData { |
| public: |
| float mVolume; |
| }; |
| |
| Mutex mLock; |
| Condition mWaitWorkCV; |
| Vector <AudioCommand *> mAudioCommands; // list of pending commands |
| ToneGenerator *mpToneGenerator; // the tone generator |
| AudioCommand mLastCommand; // last processed command (used by dump) |
| String8 mName; // string used by wake lock fo delayed commands |
| }; |
| |
| class EffectDesc { |
| public: |
| EffectDesc(const char *name, const effect_uuid_t& uuid) : |
| mName(strdup(name)), |
| mUuid(uuid) { } |
| EffectDesc(const EffectDesc& orig) : |
| mName(strdup(orig.mName)), |
| mUuid(orig.mUuid) { |
| // deep copy mParams |
| for (size_t k = 0; k < orig.mParams.size(); k++) { |
| effect_param_t *origParam = orig.mParams[k]; |
| // psize and vsize are rounded up to an int boundary for allocation |
| size_t origSize = sizeof(effect_param_t) + |
| ((origParam->psize + 3) & ~3) + |
| ((origParam->vsize + 3) & ~3); |
| effect_param_t *dupParam = (effect_param_t *) malloc(origSize); |
| memcpy(dupParam, origParam, origSize); |
| // This works because the param buffer allocation is also done by |
| // multiples of 4 bytes originally. In theory we should memcpy only |
| // the actual param size, that is without rounding vsize. |
| mParams.add(dupParam); |
| } |
| } |
| /*virtual*/ ~EffectDesc() { |
| free(mName); |
| for (size_t k = 0; k < mParams.size(); k++) { |
| free(mParams[k]); |
| } |
| } |
| char *mName; |
| effect_uuid_t mUuid; |
| Vector <effect_param_t *> mParams; |
| }; |
| |
| class InputSourceDesc { |
| public: |
| InputSourceDesc() {} |
| /*virtual*/ ~InputSourceDesc() { |
| for (size_t j = 0; j < mEffects.size(); j++) { |
| delete mEffects[j]; |
| } |
| } |
| Vector <EffectDesc *> mEffects; |
| }; |
| |
| |
| class InputDesc { |
| public: |
| InputDesc(int session) : mSessionId(session) {} |
| /*virtual*/ ~InputDesc() {} |
| const int mSessionId; |
| Vector< sp<AudioEffect> >mEffects; |
| }; |
| |
| static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1]; |
| |
| void setPreProcessorEnabled(const InputDesc *inputDesc, bool enabled); |
| status_t loadPreProcessorConfig(const char *path); |
| status_t loadEffects(cnode *root, Vector <EffectDesc *>& effects); |
| EffectDesc *loadEffect(cnode *root); |
| status_t loadInputSources(cnode *root, const Vector <EffectDesc *>& effects); |
| audio_source_t inputSourceNameToEnum(const char *name); |
| InputSourceDesc *loadInputSource(cnode *root, const Vector <EffectDesc *>& effects); |
| void loadEffectParameters(cnode *root, Vector <effect_param_t *>& params); |
| effect_param_t *loadEffectParameter(cnode *root); |
| size_t readParamValue(cnode *node, |
| char *param, |
| size_t *curSize, |
| size_t *totSize); |
| size_t growParamSize(char *param, |
| size_t size, |
| size_t *curSize, |
| size_t *totSize); |
| |
| // Internal dump utilities. |
| status_t dumpPermissionDenial(int fd); |
| |
| |
| mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing |
| // device connection state or routing |
| sp<AudioCommandThread> mAudioCommandThread; // audio commands thread |
| sp<AudioCommandThread> mTonePlaybackThread; // tone playback thread |
| struct audio_policy_device *mpAudioPolicyDev; |
| struct audio_policy *mpAudioPolicy; |
| KeyedVector< audio_source_t, InputSourceDesc* > mInputSources; |
| KeyedVector< audio_io_handle_t, InputDesc* > mInputs; |
| }; |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIOPOLICYSERVICE_H |