| /* |
| ** |
| ** Copyright 2007, 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_AUDIO_FLINGER_H |
| #define ANDROID_AUDIO_FLINGER_H |
| |
| #include <stdint.h> |
| #include <sys/types.h> |
| #include <limits.h> |
| |
| #include <common_time/cc_helper.h> |
| |
| #include <media/IAudioFlinger.h> |
| #include <media/IAudioFlingerClient.h> |
| #include <media/IAudioTrack.h> |
| #include <media/IAudioRecord.h> |
| #include <media/AudioSystem.h> |
| #include <media/AudioTrack.h> |
| |
| #include <utils/Atomic.h> |
| #include <utils/Errors.h> |
| #include <utils/threads.h> |
| #include <utils/SortedVector.h> |
| #include <utils/TypeHelpers.h> |
| #include <utils/Vector.h> |
| |
| #include <binder/BinderService.h> |
| #include <binder/MemoryDealer.h> |
| |
| #include <system/audio.h> |
| #include <hardware/audio.h> |
| #include <hardware/audio_policy.h> |
| |
| #include <media/AudioBufferProvider.h> |
| #include <media/ExtendedAudioBufferProvider.h> |
| #include "FastMixer.h" |
| #include <media/nbaio/NBAIO.h> |
| #include "AudioWatchdog.h" |
| |
| #include <powermanager/IPowerManager.h> |
| |
| #include <media/nbaio/NBLog.h> |
| |
| namespace android { |
| |
| class audio_track_cblk_t; |
| class effect_param_cblk_t; |
| class AudioMixer; |
| class AudioBuffer; |
| class AudioResampler; |
| class FastMixer; |
| class ServerProxy; |
| |
| // ---------------------------------------------------------------------------- |
| |
| // AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. |
| // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. |
| // Adding full support for > 2 channel capture or playback would require more than simply changing |
| // this #define. There is an independent hard-coded upper limit in AudioMixer; |
| // removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. |
| // The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. |
| // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. |
| #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 |
| |
| static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); |
| |
| #define MAX_GAIN 4096.0f |
| #define MAX_GAIN_INT 0x1000 |
| |
| #define INCLUDING_FROM_AUDIOFLINGER_H |
| |
| class AudioFlinger : |
| public BinderService<AudioFlinger>, |
| public BnAudioFlinger |
| { |
| friend class BinderService<AudioFlinger>; // for AudioFlinger() |
| public: |
| static const char* getServiceName() { return "media.audio_flinger"; } |
| |
| virtual status_t dump(int fd, const Vector<String16>& args); |
| |
| // IAudioFlinger interface, in binder opcode order |
| virtual sp<IAudioTrack> createTrack( |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| IAudioFlinger::track_flags_t *flags, |
| const sp<IMemory>& sharedBuffer, |
| audio_io_handle_t output, |
| pid_t tid, |
| int *sessionId, |
| status_t *status); |
| |
| virtual sp<IAudioRecord> openRecord( |
| audio_io_handle_t input, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| IAudioFlinger::track_flags_t flags, |
| pid_t tid, |
| int *sessionId, |
| status_t *status); |
| |
| virtual uint32_t sampleRate(audio_io_handle_t output) const; |
| virtual int channelCount(audio_io_handle_t output) const; |
| virtual audio_format_t format(audio_io_handle_t output) const; |
| virtual size_t frameCount(audio_io_handle_t output) const; |
| virtual uint32_t latency(audio_io_handle_t output) const; |
| |
| virtual status_t setMasterVolume(float value); |
| virtual status_t setMasterMute(bool muted); |
| |
| virtual float masterVolume() const; |
| virtual bool masterMute() const; |
| |
| virtual status_t setStreamVolume(audio_stream_type_t stream, float value, |
| audio_io_handle_t output); |
| virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); |
| |
| virtual float streamVolume(audio_stream_type_t stream, |
| audio_io_handle_t output) const; |
| virtual bool streamMute(audio_stream_type_t stream) const; |
| |
| virtual status_t setMode(audio_mode_t mode); |
| |
| virtual status_t setMicMute(bool state); |
| virtual bool getMicMute() const; |
| |
| virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); |
| virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; |
| |
| virtual void registerClient(const sp<IAudioFlingerClient>& client); |
| |
| virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, |
| audio_channel_mask_t channelMask) const; |
| |
| virtual audio_io_handle_t openOutput(audio_module_handle_t module, |
| audio_devices_t *pDevices, |
| uint32_t *pSamplingRate, |
| audio_format_t *pFormat, |
| audio_channel_mask_t *pChannelMask, |
| uint32_t *pLatencyMs, |
| audio_output_flags_t flags); |
| |
| virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, |
| audio_io_handle_t output2); |
| |
| virtual status_t closeOutput(audio_io_handle_t output); |
| |
| virtual status_t suspendOutput(audio_io_handle_t output); |
| |
| virtual status_t restoreOutput(audio_io_handle_t output); |
| |
| virtual audio_io_handle_t openInput(audio_module_handle_t module, |
| audio_devices_t *pDevices, |
| uint32_t *pSamplingRate, |
| audio_format_t *pFormat, |
| audio_channel_mask_t *pChannelMask); |
| |
| virtual status_t closeInput(audio_io_handle_t input); |
| |
| virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); |
| |
| virtual status_t setVoiceVolume(float volume); |
| |
| virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, |
| audio_io_handle_t output) const; |
| |
| virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; |
| |
| virtual int newAudioSessionId(); |
| |
| virtual void acquireAudioSessionId(int audioSession); |
| |
| virtual void releaseAudioSessionId(int audioSession); |
| |
| virtual status_t queryNumberEffects(uint32_t *numEffects) const; |
| |
| virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; |
| |
| virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, |
| effect_descriptor_t *descriptor) const; |
| |
| virtual sp<IEffect> createEffect( |
| effect_descriptor_t *pDesc, |
| const sp<IEffectClient>& effectClient, |
| int32_t priority, |
| audio_io_handle_t io, |
| int sessionId, |
| status_t *status, |
| int *id, |
| int *enabled); |
| |
| virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, |
| audio_io_handle_t dstOutput); |
| |
| virtual audio_module_handle_t loadHwModule(const char *name); |
| |
| virtual uint32_t getPrimaryOutputSamplingRate(); |
| virtual size_t getPrimaryOutputFrameCount(); |
| |
| virtual status_t onTransact( |
| uint32_t code, |
| const Parcel& data, |
| Parcel* reply, |
| uint32_t flags); |
| |
| // end of IAudioFlinger interface |
| |
| sp<NBLog::Writer> newWriter_l(size_t size, const char *name); |
| void unregisterWriter(const sp<NBLog::Writer>& writer); |
| private: |
| static const size_t kLogMemorySize = 10 * 1024; |
| sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled |
| public: |
| |
| class SyncEvent; |
| |
| typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; |
| |
| class SyncEvent : public RefBase { |
| public: |
| SyncEvent(AudioSystem::sync_event_t type, |
| int triggerSession, |
| int listenerSession, |
| sync_event_callback_t callBack, |
| void *cookie) |
| : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), |
| mCallback(callBack), mCookie(cookie) |
| {} |
| |
| virtual ~SyncEvent() {} |
| |
| void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } |
| bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } |
| void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } |
| AudioSystem::sync_event_t type() const { return mType; } |
| int triggerSession() const { return mTriggerSession; } |
| int listenerSession() const { return mListenerSession; } |
| void *cookie() const { return mCookie; } |
| |
| private: |
| const AudioSystem::sync_event_t mType; |
| const int mTriggerSession; |
| const int mListenerSession; |
| sync_event_callback_t mCallback; |
| void * const mCookie; |
| mutable Mutex mLock; |
| }; |
| |
| sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, |
| int triggerSession, |
| int listenerSession, |
| sync_event_callback_t callBack, |
| void *cookie); |
| |
| private: |
| class AudioHwDevice; // fwd declaration for findSuitableHwDev_l |
| |
| audio_mode_t getMode() const { return mMode; } |
| |
| bool btNrecIsOff() const { return mBtNrecIsOff; } |
| |
| AudioFlinger(); |
| virtual ~AudioFlinger(); |
| |
| // call in any IAudioFlinger method that accesses mPrimaryHardwareDev |
| status_t initCheck() const { return mPrimaryHardwareDev == NULL ? |
| NO_INIT : NO_ERROR; } |
| |
| // RefBase |
| virtual void onFirstRef(); |
| |
| AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, |
| audio_devices_t devices); |
| void purgeStaleEffects_l(); |
| |
| // standby delay for MIXER and DUPLICATING playback threads is read from property |
| // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs |
| static nsecs_t mStandbyTimeInNsecs; |
| |
| // incremented by 2 when screen state changes, bit 0 == 1 means "off" |
| // AudioFlinger::setParameters() updates, other threads read w/o lock |
| static uint32_t mScreenState; |
| |
| // Internal dump utilities. |
| static const int kDumpLockRetries = 50; |
| static const int kDumpLockSleepUs = 20000; |
| static bool dumpTryLock(Mutex& mutex); |
| void dumpPermissionDenial(int fd, const Vector<String16>& args); |
| void dumpClients(int fd, const Vector<String16>& args); |
| void dumpInternals(int fd, const Vector<String16>& args); |
| |
| // --- Client --- |
| class Client : public RefBase { |
| public: |
| Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); |
| virtual ~Client(); |
| sp<MemoryDealer> heap() const; |
| pid_t pid() const { return mPid; } |
| sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } |
| |
| bool reserveTimedTrack(); |
| void releaseTimedTrack(); |
| |
| private: |
| Client(const Client&); |
| Client& operator = (const Client&); |
| const sp<AudioFlinger> mAudioFlinger; |
| const sp<MemoryDealer> mMemoryDealer; |
| const pid_t mPid; |
| |
| Mutex mTimedTrackLock; |
| int mTimedTrackCount; |
| }; |
| |
| // --- Notification Client --- |
| class NotificationClient : public IBinder::DeathRecipient { |
| public: |
| NotificationClient(const sp<AudioFlinger>& audioFlinger, |
| const sp<IAudioFlingerClient>& client, |
| pid_t pid); |
| virtual ~NotificationClient(); |
| |
| sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } |
| |
| // IBinder::DeathRecipient |
| virtual void binderDied(const wp<IBinder>& who); |
| |
| private: |
| NotificationClient(const NotificationClient&); |
| NotificationClient& operator = (const NotificationClient&); |
| |
| const sp<AudioFlinger> mAudioFlinger; |
| const pid_t mPid; |
| const sp<IAudioFlingerClient> mAudioFlingerClient; |
| }; |
| |
| class TrackHandle; |
| class RecordHandle; |
| class RecordThread; |
| class PlaybackThread; |
| class MixerThread; |
| class DirectOutputThread; |
| class DuplicatingThread; |
| class Track; |
| class RecordTrack; |
| class EffectModule; |
| class EffectHandle; |
| class EffectChain; |
| struct AudioStreamOut; |
| struct AudioStreamIn; |
| |
| struct stream_type_t { |
| stream_type_t() |
| : volume(1.0f), |
| mute(false) |
| { |
| } |
| float volume; |
| bool mute; |
| }; |
| |
| // --- PlaybackThread --- |
| |
| #include "Threads.h" |
| |
| #include "Effects.h" |
| |
| // server side of the client's IAudioTrack |
| class TrackHandle : public android::BnAudioTrack { |
| public: |
| TrackHandle(const sp<PlaybackThread::Track>& track); |
| virtual ~TrackHandle(); |
| virtual sp<IMemory> getCblk() const; |
| virtual status_t start(); |
| virtual void stop(); |
| virtual void flush(); |
| virtual void pause(); |
| virtual status_t attachAuxEffect(int effectId); |
| virtual status_t allocateTimedBuffer(size_t size, |
| sp<IMemory>* buffer); |
| virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, |
| int64_t pts); |
| virtual status_t setMediaTimeTransform(const LinearTransform& xform, |
| int target); |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| private: |
| const sp<PlaybackThread::Track> mTrack; |
| }; |
| |
| // server side of the client's IAudioRecord |
| class RecordHandle : public android::BnAudioRecord { |
| public: |
| RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); |
| virtual ~RecordHandle(); |
| virtual sp<IMemory> getCblk() const; |
| virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); |
| virtual void stop(); |
| virtual status_t onTransact( |
| uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); |
| private: |
| const sp<RecordThread::RecordTrack> mRecordTrack; |
| |
| // for use from destructor |
| void stop_nonvirtual(); |
| }; |
| |
| PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; |
| MixerThread *checkMixerThread_l(audio_io_handle_t output) const; |
| RecordThread *checkRecordThread_l(audio_io_handle_t input) const; |
| // no range check, AudioFlinger::mLock held |
| bool streamMute_l(audio_stream_type_t stream) const |
| { return mStreamTypes[stream].mute; } |
| // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held |
| float streamVolume_l(audio_stream_type_t stream) const |
| { return mStreamTypes[stream].volume; } |
| void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); |
| |
| // allocate an audio_io_handle_t, session ID, or effect ID |
| uint32_t nextUniqueId(); |
| |
| status_t moveEffectChain_l(int sessionId, |
| PlaybackThread *srcThread, |
| PlaybackThread *dstThread, |
| bool reRegister); |
| // return thread associated with primary hardware device, or NULL |
| PlaybackThread *primaryPlaybackThread_l() const; |
| audio_devices_t primaryOutputDevice_l() const; |
| |
| sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); |
| |
| |
| void removeClient_l(pid_t pid); |
| void removeNotificationClient(pid_t pid); |
| |
| class AudioHwDevice { |
| public: |
| enum Flags { |
| AHWD_CAN_SET_MASTER_VOLUME = 0x1, |
| AHWD_CAN_SET_MASTER_MUTE = 0x2, |
| }; |
| |
| AudioHwDevice(const char *moduleName, |
| audio_hw_device_t *hwDevice, |
| Flags flags) |
| : mModuleName(strdup(moduleName)) |
| , mHwDevice(hwDevice) |
| , mFlags(flags) { } |
| /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } |
| |
| bool canSetMasterVolume() const { |
| return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); |
| } |
| |
| bool canSetMasterMute() const { |
| return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); |
| } |
| |
| const char *moduleName() const { return mModuleName; } |
| audio_hw_device_t *hwDevice() const { return mHwDevice; } |
| private: |
| const char * const mModuleName; |
| audio_hw_device_t * const mHwDevice; |
| Flags mFlags; |
| }; |
| |
| // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. |
| // For emphasis, we could also make all pointers to them be "const *", |
| // but that would clutter the code unnecessarily. |
| |
| struct AudioStreamOut { |
| AudioHwDevice* const audioHwDev; |
| audio_stream_out_t* const stream; |
| |
| audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } |
| |
| AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : |
| audioHwDev(dev), stream(out) {} |
| }; |
| |
| struct AudioStreamIn { |
| AudioHwDevice* const audioHwDev; |
| audio_stream_in_t* const stream; |
| |
| audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } |
| |
| AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : |
| audioHwDev(dev), stream(in) {} |
| }; |
| |
| // for mAudioSessionRefs only |
| struct AudioSessionRef { |
| AudioSessionRef(int sessionid, pid_t pid) : |
| mSessionid(sessionid), mPid(pid), mCnt(1) {} |
| const int mSessionid; |
| const pid_t mPid; |
| int mCnt; |
| }; |
| |
| mutable Mutex mLock; |
| |
| DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() |
| |
| mutable Mutex mHardwareLock; |
| // NOTE: If both mLock and mHardwareLock mutexes must be held, |
| // always take mLock before mHardwareLock |
| |
| // These two fields are immutable after onFirstRef(), so no lock needed to access |
| AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL |
| DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; |
| |
| // for dump, indicates which hardware operation is currently in progress (but not stream ops) |
| enum hardware_call_state { |
| AUDIO_HW_IDLE = 0, // no operation in progress |
| AUDIO_HW_INIT, // init_check |
| AUDIO_HW_OUTPUT_OPEN, // open_output_stream |
| AUDIO_HW_OUTPUT_CLOSE, // unused |
| AUDIO_HW_INPUT_OPEN, // unused |
| AUDIO_HW_INPUT_CLOSE, // unused |
| AUDIO_HW_STANDBY, // unused |
| AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume |
| AUDIO_HW_GET_ROUTING, // unused |
| AUDIO_HW_SET_ROUTING, // unused |
| AUDIO_HW_GET_MODE, // unused |
| AUDIO_HW_SET_MODE, // set_mode |
| AUDIO_HW_GET_MIC_MUTE, // get_mic_mute |
| AUDIO_HW_SET_MIC_MUTE, // set_mic_mute |
| AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume |
| AUDIO_HW_SET_PARAMETER, // set_parameters |
| AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size |
| AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume |
| AUDIO_HW_GET_PARAMETER, // get_parameters |
| AUDIO_HW_SET_MASTER_MUTE, // set_master_mute |
| AUDIO_HW_GET_MASTER_MUTE, // get_master_mute |
| }; |
| |
| mutable hardware_call_state mHardwareStatus; // for dump only |
| |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; |
| stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; |
| |
| // member variables below are protected by mLock |
| float mMasterVolume; |
| bool mMasterMute; |
| // end of variables protected by mLock |
| |
| DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; |
| |
| DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; |
| volatile int32_t mNextUniqueId; // updated by android_atomic_inc |
| audio_mode_t mMode; |
| bool mBtNrecIsOff; |
| |
| // protected by mLock |
| Vector<AudioSessionRef*> mAudioSessionRefs; |
| |
| float masterVolume_l() const; |
| bool masterMute_l() const; |
| audio_module_handle_t loadHwModule_l(const char *name); |
| |
| Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session |
| // to be created |
| |
| private: |
| sp<Client> registerPid_l(pid_t pid); // always returns non-0 |
| |
| // for use from destructor |
| status_t closeOutput_nonvirtual(audio_io_handle_t output); |
| status_t closeInput_nonvirtual(audio_io_handle_t input); |
| |
| // do not use #ifdef here, since AudioFlinger.h is included by more than one module |
| //#ifdef TEE_SINK |
| // all record threads serially share a common tee sink, which is re-created on format change |
| sp<NBAIO_Sink> mRecordTeeSink; |
| sp<NBAIO_Source> mRecordTeeSource; |
| //#endif |
| |
| public: |
| |
| #ifdef TEE_SINK |
| // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file |
| static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); |
| |
| // whether tee sink is enabled by property |
| static bool mTeeSinkInputEnabled; |
| static bool mTeeSinkOutputEnabled; |
| static bool mTeeSinkTrackEnabled; |
| |
| // runtime configured size of each tee sink pipe, in frames |
| static size_t mTeeSinkInputFrames; |
| static size_t mTeeSinkOutputFrames; |
| static size_t mTeeSinkTrackFrames; |
| |
| // compile-time default size of tee sink pipes, in frames |
| // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes |
| static const size_t kTeeSinkInputFramesDefault = 0x200000; |
| static const size_t kTeeSinkOutputFramesDefault = 0x200000; |
| static const size_t kTeeSinkTrackFramesDefault = 0x1000; |
| #endif |
| |
| }; |
| |
| #undef INCLUDING_FROM_AUDIOFLINGER_H |
| |
| // ---------------------------------------------------------------------------- |
| |
| }; // namespace android |
| |
| #endif // ANDROID_AUDIO_FLINGER_H |
