/* ** ** 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 "Configuration.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "FastMixer.h" #include #include "AudioWatchdog.h" #include #include #include namespace android { struct audio_track_cblk_t; struct 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, public BnAudioFlinger { friend class BinderService; // for AudioFlinger() public: static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } virtual status_t dump(int fd, const Vector& args); // IAudioFlinger interface, in binder opcode order virtual sp createTrack( audio_stream_type_t streamType, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, IAudioFlinger::track_flags_t *flags, const sp& sharedBuffer, audio_io_handle_t output, pid_t tid, int *sessionId, int clientUid, status_t *status /*non-NULL*/); virtual sp openRecord( audio_io_handle_t input, uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, IAudioFlinger::track_flags_t *flags, pid_t tid, int *sessionId, sp& cblk, sp& buffers, status_t *status /*non-NULL*/); 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& 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, const audio_offload_info_t *offloadInfo); 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 invalidateStream(audio_stream_type_t stream); virtual status_t setVoiceVolume(float volume); virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_io_handle_t output) const; virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; virtual int newAudioSessionId(); virtual void acquireAudioSessionId(int audioSession, pid_t pid); virtual void releaseAudioSessionId(int audioSession, pid_t pid); 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 createEffect( effect_descriptor_t *pDesc, const sp& effectClient, int32_t priority, audio_io_handle_t io, int sessionId, status_t *status /*non-NULL*/, 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 setLowRamDevice(bool isLowRamDevice); virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); // end of IAudioFlinger interface sp newWriter_l(size_t size, const char *name); void unregisterWriter(const sp& writer); private: static const size_t kLogMemorySize = 40 * 1024; sp mLogMemoryDealer; // == 0 when NBLog is disabled // When a log writer is unregistered, it is done lazily so that media.log can continue to see it // for as long as possible. The memory is only freed when it is needed for another log writer. Vector< sp > mUnregisteredWriters; Mutex mUnregisteredWritersLock; public: class SyncEvent; typedef void (*sync_event_callback_t)(const wp& event) ; class SyncEvent : public RefBase { public: SyncEvent(AudioSystem::sync_event_t type, int triggerSession, int listenerSession, sync_event_callback_t callBack, wp 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; } wp cookie() const { return mCookie; } private: const AudioSystem::sync_event_t mType; const int mTriggerSession; const int mListenerSession; sync_event_callback_t mCallback; const wp mCookie; mutable Mutex mLock; }; sp createSyncEvent(AudioSystem::sync_event_t type, int triggerSession, int listenerSession, sync_event_callback_t callBack, wp cookie); private: class AudioHwDevice; // fwd declaration for findSuitableHwDev_l audio_mode_t getMode() const { return mMode; } bool btNrecIsOff() const { return mBtNrecIsOff; } AudioFlinger() ANDROID_API; 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& args); void dumpClients(int fd, const Vector& args); void dumpInternals(int fd, const Vector& args); // --- Client --- class Client : public RefBase { public: Client(const sp& audioFlinger, pid_t pid); virtual ~Client(); sp heap() const; pid_t pid() const { return mPid; } sp audioFlinger() const { return mAudioFlinger; } bool reserveTimedTrack(); void releaseTimedTrack(); private: Client(const Client&); Client& operator = (const Client&); const sp mAudioFlinger; const sp mMemoryDealer; const pid_t mPid; Mutex mTimedTrackLock; int mTimedTrackCount; }; // --- Notification Client --- class NotificationClient : public IBinder::DeathRecipient { public: NotificationClient(const sp& audioFlinger, const sp& client, pid_t pid); virtual ~NotificationClient(); sp audioFlingerClient() const { return mAudioFlingerClient; } // IBinder::DeathRecipient virtual void binderDied(const wp& who); private: NotificationClient(const NotificationClient&); NotificationClient& operator = (const NotificationClient&); const sp mAudioFlinger; const pid_t mPid; const sp mAudioFlingerClient; }; class TrackHandle; class RecordHandle; class RecordThread; class PlaybackThread; class MixerThread; class DirectOutputThread; class OffloadThread; class DuplicatingThread; class AsyncCallbackThread; 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& track); virtual ~TrackHandle(); virtual sp 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* buffer); virtual status_t queueTimedBuffer(const sp& buffer, int64_t pts); virtual status_t setMediaTimeTransform(const LinearTransform& xform, int target); virtual status_t setParameters(const String8& keyValuePairs); virtual status_t getTimestamp(AudioTimestamp& timestamp); virtual void signal(); // signal playback thread for a change in control block virtual status_t onTransact( uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); private: const sp mTrack; }; // server side of the client's IAudioRecord class RecordHandle : public android::BnAudioRecord { public: RecordHandle(const sp& recordTrack); virtual ~RecordHandle(); 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 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(int event, audio_io_handle_t ioHandle, const void *param2); // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. // They all share the same ID space, but the namespaces are actually independent // because there are separate KeyedVectors for each kind of ID. // The return value is uint32_t, but is cast to signed for some IDs. // FIXME This API does not handle rollover to zero (for unsigned IDs), // or from positive to negative (for signed IDs). // Thus it may fail by returning an ID of the wrong sign, // or by returning a non-unique 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 getEffectThread_l(int sessionId, int EffectId); void removeClient_l(pid_t pid); void removeNotificationClient(pid_t pid); bool isNonOffloadableGlobalEffectEnabled_l(); void onNonOffloadableGlobalEffectEnable(); 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; const 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; const audio_output_flags_t flags; audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : audioHwDev(dev), stream(out), flags(flags) {} }; 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; // protects mClients and mNotificationClients. // must be locked after mLock and ThreadBase::mLock if both must be locked // avoids acquiring AudioFlinger::mLock from inside thread loop. mutable Mutex mClientLock; // protected by mClientLock DefaultKeyedVector< pid_t, wp > 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 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 > 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 > mRecordThreads; // protected by mClientLock DefaultKeyedVector< pid_t, sp > mNotificationClients; volatile int32_t mNextUniqueId; // updated by android_atomic_inc // nextUniqueId() returns uint32_t, but this is declared int32_t // because the atomic operations require an int32_t audio_mode_t mMode; bool mBtNrecIsOff; // protected by mLock Vector mAudioSessionRefs; float masterVolume_l() const; bool masterMute_l() const; audio_module_handle_t loadHwModule_l(const char *name); Vector < sp > mPendingSyncEvents; // sync events awaiting for a session // to be created private: sp registerPid(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); #ifdef TEE_SINK // all record threads serially share a common tee sink, which is re-created on format change sp mRecordTeeSink; sp 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& 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 = 0x200000; #endif // This method reads from a variable without mLock, but the variable is updated under mLock. So // we might read a stale value, or a value that's inconsistent with respect to other variables. // In this case, it's safe because the return value isn't used for making an important decision. // The reason we don't want to take mLock is because it could block the caller for a long time. bool isLowRamDevice() const { return mIsLowRamDevice; } private: bool mIsLowRamDevice; bool mIsDeviceTypeKnown; nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled }; #undef INCLUDING_FROM_AUDIOFLINGER_H const char *formatToString(audio_format_t format); // ---------------------------------------------------------------------------- }; // namespace android #endif // ANDROID_AUDIO_FLINGER_H