/* ** ** Copyright 2012, 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 INCLUDING_FROM_AUDIOFLINGER_H #error This header file should only be included from AudioFlinger.h #endif class ThreadBase : public Thread { public: #include "TrackBase.h" enum type_t { MIXER, // Thread class is MixerThread DIRECT, // Thread class is DirectOutputThread DUPLICATING, // Thread class is DuplicatingThread RECORD, // Thread class is RecordThread OFFLOAD, // Thread class is OffloadThread MMAP // control thread for MMAP stream // If you add any values here, also update ThreadBase::threadTypeToString() }; static const char *threadTypeToString(type_t type); ThreadBase(const sp& audioFlinger, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady); virtual ~ThreadBase(); virtual status_t readyToRun(); void clearPowerManager(); // base for record and playback enum { CFG_EVENT_IO, CFG_EVENT_PRIO, CFG_EVENT_SET_PARAMETER, CFG_EVENT_CREATE_AUDIO_PATCH, CFG_EVENT_RELEASE_AUDIO_PATCH, }; class ConfigEventData: public RefBase { public: virtual ~ConfigEventData() {} virtual void dump(char *buffer, size_t size) = 0; protected: ConfigEventData() {} }; // Config event sequence by client if status needed (e.g binder thread calling setParameters()): // 1. create SetParameterConfigEvent. This sets mWaitStatus in config event // 2. Lock mLock // 3. Call sendConfigEvent_l(): Append to mConfigEvents and mWaitWorkCV.signal // 4. sendConfigEvent_l() reads status from event->mStatus; // 5. sendConfigEvent_l() returns status // 6. Unlock // // Parameter sequence by server: threadLoop calling processConfigEvents_l(): // 1. Lock mLock // 2. If there is an entry in mConfigEvents proceed ... // 3. Read first entry in mConfigEvents // 4. Remove first entry from mConfigEvents // 5. Process // 6. Set event->mStatus // 7. event->mCond.signal // 8. Unlock class ConfigEvent: public RefBase { public: virtual ~ConfigEvent() {} void dump(char *buffer, size_t size) { mData->dump(buffer, size); } const int mType; // event type e.g. CFG_EVENT_IO Mutex mLock; // mutex associated with mCond Condition mCond; // condition for status return status_t mStatus; // status communicated to sender bool mWaitStatus; // true if sender is waiting for status bool mRequiresSystemReady; // true if must wait for system ready to enter event queue sp mData; // event specific parameter data protected: explicit ConfigEvent(int type, bool requiresSystemReady = false) : mType(type), mStatus(NO_ERROR), mWaitStatus(false), mRequiresSystemReady(requiresSystemReady), mData(NULL) {} }; class IoConfigEventData : public ConfigEventData { public: IoConfigEventData(audio_io_config_event event, pid_t pid, audio_port_handle_t portId) : mEvent(event), mPid(pid), mPortId(portId) {} virtual void dump(char *buffer, size_t size) { snprintf(buffer, size, "IO event: event %d\n", mEvent); } const audio_io_config_event mEvent; const pid_t mPid; const audio_port_handle_t mPortId; }; class IoConfigEvent : public ConfigEvent { public: IoConfigEvent(audio_io_config_event event, pid_t pid, audio_port_handle_t portId) : ConfigEvent(CFG_EVENT_IO) { mData = new IoConfigEventData(event, pid, portId); } virtual ~IoConfigEvent() {} }; class PrioConfigEventData : public ConfigEventData { public: PrioConfigEventData(pid_t pid, pid_t tid, int32_t prio, bool forApp) : mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {} virtual void dump(char *buffer, size_t size) { snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n", mPid, mTid, mPrio, mForApp); } const pid_t mPid; const pid_t mTid; const int32_t mPrio; const bool mForApp; }; class PrioConfigEvent : public ConfigEvent { public: PrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) : ConfigEvent(CFG_EVENT_PRIO, true) { mData = new PrioConfigEventData(pid, tid, prio, forApp); } virtual ~PrioConfigEvent() {} }; class SetParameterConfigEventData : public ConfigEventData { public: explicit SetParameterConfigEventData(String8 keyValuePairs) : mKeyValuePairs(keyValuePairs) {} virtual void dump(char *buffer, size_t size) { snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string()); } const String8 mKeyValuePairs; }; class SetParameterConfigEvent : public ConfigEvent { public: explicit SetParameterConfigEvent(String8 keyValuePairs) : ConfigEvent(CFG_EVENT_SET_PARAMETER) { mData = new SetParameterConfigEventData(keyValuePairs); mWaitStatus = true; } virtual ~SetParameterConfigEvent() {} }; class CreateAudioPatchConfigEventData : public ConfigEventData { public: CreateAudioPatchConfigEventData(const struct audio_patch patch, audio_patch_handle_t handle) : mPatch(patch), mHandle(handle) {} virtual void dump(char *buffer, size_t size) { snprintf(buffer, size, "Patch handle: %u\n", mHandle); } const struct audio_patch mPatch; audio_patch_handle_t mHandle; }; class CreateAudioPatchConfigEvent : public ConfigEvent { public: CreateAudioPatchConfigEvent(const struct audio_patch patch, audio_patch_handle_t handle) : ConfigEvent(CFG_EVENT_CREATE_AUDIO_PATCH) { mData = new CreateAudioPatchConfigEventData(patch, handle); mWaitStatus = true; } virtual ~CreateAudioPatchConfigEvent() {} }; class ReleaseAudioPatchConfigEventData : public ConfigEventData { public: explicit ReleaseAudioPatchConfigEventData(const audio_patch_handle_t handle) : mHandle(handle) {} virtual void dump(char *buffer, size_t size) { snprintf(buffer, size, "Patch handle: %u\n", mHandle); } audio_patch_handle_t mHandle; }; class ReleaseAudioPatchConfigEvent : public ConfigEvent { public: explicit ReleaseAudioPatchConfigEvent(const audio_patch_handle_t handle) : ConfigEvent(CFG_EVENT_RELEASE_AUDIO_PATCH) { mData = new ReleaseAudioPatchConfigEventData(handle); mWaitStatus = true; } virtual ~ReleaseAudioPatchConfigEvent() {} }; class PMDeathRecipient : public IBinder::DeathRecipient { public: explicit PMDeathRecipient(const wp& thread) : mThread(thread) {} virtual ~PMDeathRecipient() {} // IBinder::DeathRecipient virtual void binderDied(const wp& who); private: DISALLOW_COPY_AND_ASSIGN(PMDeathRecipient); wp mThread; }; virtual status_t initCheck() const = 0; // static externally-visible type_t type() const { return mType; } bool isDuplicating() const { return (mType == DUPLICATING); } audio_io_handle_t id() const { return mId;} // dynamic externally-visible uint32_t sampleRate() const { return mSampleRate; } audio_channel_mask_t channelMask() const { return mChannelMask; } audio_format_t format() const { return mHALFormat; } uint32_t channelCount() const { return mChannelCount; } // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects, // and returns the [normal mix] buffer's frame count. virtual size_t frameCount() const = 0; // Return's the HAL's frame count i.e. fast mixer buffer size. size_t frameCountHAL() const { return mFrameCount; } size_t frameSize() const { return mFrameSize; } // Should be "virtual status_t requestExitAndWait()" and override same // method in Thread, but Thread::requestExitAndWait() is not yet virtual. void exit(); virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) = 0; virtual status_t setParameters(const String8& keyValuePairs); virtual String8 getParameters(const String8& keys) = 0; virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0; // sendConfigEvent_l() must be called with ThreadBase::mLock held // Can temporarily release the lock if waiting for a reply from // processConfigEvents_l(). status_t sendConfigEvent_l(sp& event); void sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE); void sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE); void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp); void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp); status_t sendSetParameterConfigEvent_l(const String8& keyValuePair); status_t sendCreateAudioPatchConfigEvent(const struct audio_patch *patch, audio_patch_handle_t *handle); status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle); void processConfigEvents_l(); virtual void cacheParameters_l() = 0; virtual status_t createAudioPatch_l(const struct audio_patch *patch, audio_patch_handle_t *handle) = 0; virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle) = 0; virtual void toAudioPortConfig(struct audio_port_config *config) = 0; // see note at declaration of mStandby, mOutDevice and mInDevice bool standby() const { return mStandby; } audio_devices_t outDevice() const { return mOutDevice; } audio_devices_t inDevice() const { return mInDevice; } audio_devices_t getDevice() const { return isOutput() ? mOutDevice : mInDevice; } virtual bool isOutput() const = 0; virtual sp stream() const = 0; sp createEffect_l( const sp& client, const sp& effectClient, int32_t priority, audio_session_t sessionId, effect_descriptor_t *desc, int *enabled, status_t *status /*non-NULL*/, bool pinned); // return values for hasAudioSession (bit field) enum effect_state { EFFECT_SESSION = 0x1, // the audio session corresponds to at least one // effect TRACK_SESSION = 0x2, // the audio session corresponds to at least one // track FAST_SESSION = 0x4 // the audio session corresponds to at least one // fast track }; // get effect chain corresponding to session Id. sp getEffectChain(audio_session_t sessionId); // same as getEffectChain() but must be called with ThreadBase mutex locked sp getEffectChain_l(audio_session_t sessionId) const; std::vector getEffectIds_l(audio_session_t sessionId); // add an effect chain to the chain list (mEffectChains) virtual status_t addEffectChain_l(const sp& chain) = 0; // remove an effect chain from the chain list (mEffectChains) virtual size_t removeEffectChain_l(const sp& chain) = 0; // lock all effect chains Mutexes. Must be called before releasing the // ThreadBase mutex before processing the mixer and effects. This guarantees the // integrity of the chains during the process. // Also sets the parameter 'effectChains' to current value of mEffectChains. void lockEffectChains_l(Vector< sp >& effectChains); // unlock effect chains after process void unlockEffectChains(const Vector< sp >& effectChains); // get a copy of mEffectChains vector Vector< sp > getEffectChains_l() const { return mEffectChains; }; // set audio mode to all effect chains void setMode(audio_mode_t mode); // get effect module with corresponding ID on specified audio session sp getEffect(audio_session_t sessionId, int effectId); sp getEffect_l(audio_session_t sessionId, int effectId); // add and effect module. Also creates the effect chain is none exists for // the effects audio session. Only called in a context of moving an effect // from one thread to another status_t addEffect_l(const sp< EffectModule>& effect); // remove and effect module. Also removes the effect chain is this was the last // effect void removeEffect_l(const sp< EffectModule>& effect, bool release = false); // disconnect an effect handle from module and destroy module if last handle void disconnectEffectHandle(EffectHandle *handle, bool unpinIfLast); // detach all tracks connected to an auxiliary effect virtual void detachAuxEffect_l(int effectId __unused) {} // returns a combination of: // - EFFECT_SESSION if effects on this audio session exist in one chain // - TRACK_SESSION if tracks on this audio session exist // - FAST_SESSION if fast tracks on this audio session exist virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0; uint32_t hasAudioSession(audio_session_t sessionId) const { Mutex::Autolock _l(mLock); return hasAudioSession_l(sessionId); } template uint32_t hasAudioSession_l(audio_session_t sessionId, const T& tracks) const { uint32_t result = 0; if (getEffectChain_l(sessionId) != 0) { result = EFFECT_SESSION; } for (size_t i = 0; i < tracks.size(); ++i) { const sp& track = tracks[i]; if (sessionId == track->sessionId() && !track->isInvalid() // not yet removed from tracks. && !track->isTerminated()) { result |= TRACK_SESSION; if (track->isFastTrack()) { result |= FAST_SESSION; // caution, only represents first track. } break; } } return result; } // the value returned by default implementation is not important as the // strategy is only meaningful for PlaybackThread which implements this method virtual uint32_t getStrategyForSession_l(audio_session_t sessionId __unused) { return 0; } // check if some effects must be suspended/restored when an effect is enabled // or disabled void checkSuspendOnEffectEnabled(const sp& effect, bool enabled, audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); void checkSuspendOnEffectEnabled_l(const sp& effect, bool enabled, audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX); virtual status_t setSyncEvent(const sp& event) = 0; virtual bool isValidSyncEvent(const sp& event) const = 0; // Return a reference to a per-thread heap which can be used to allocate IMemory // objects that will be read-only to client processes, read/write to mediaserver, // and shared by all client processes of the thread. // The heap is per-thread rather than common across all threads, because // clients can't be trusted not to modify the offset of the IMemory they receive. // If a thread does not have such a heap, this method returns 0. virtual sp readOnlyHeap() const { return 0; } virtual sp pipeMemory() const { return 0; } void systemReady(); // checkEffectCompatibility_l() must be called with ThreadBase::mLock held virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, audio_session_t sessionId) = 0; void broadcast_l(); virtual bool isTimestampCorrectionEnabled() const { return false; } bool isMsdDevice() const { return mIsMsdDevice; } void dump(int fd, const Vector& args); // deliver stats to mediametrics. void sendStatistics(bool force); mutable Mutex mLock; protected: // entry describing an effect being suspended in mSuspendedSessions keyed vector class SuspendedSessionDesc : public RefBase { public: SuspendedSessionDesc() : mRefCount(0) {} int mRefCount; // number of active suspend requests effect_uuid_t mType; // effect type UUID }; void acquireWakeLock(); virtual void acquireWakeLock_l(); void releaseWakeLock(); void releaseWakeLock_l(); void updateWakeLockUids_l(const SortedVector &uids); void getPowerManager_l(); // suspend or restore effects of the specified type (or all if type is NULL) // on a given session. The number of suspend requests is counted and restore // occurs when all suspend requests are cancelled. void setEffectSuspended_l(const effect_uuid_t *type, bool suspend, audio_session_t sessionId); // updated mSuspendedSessions when an effect is suspended or restored void updateSuspendedSessions_l(const effect_uuid_t *type, bool suspend, audio_session_t sessionId); // check if some effects must be suspended when an effect chain is added void checkSuspendOnAddEffectChain_l(const sp& chain); // sends the metadata of the active tracks to the HAL virtual void updateMetadata_l() = 0; String16 getWakeLockTag(); virtual void preExit() { } virtual void setMasterMono_l(bool mono __unused) { } virtual bool requireMonoBlend() { return false; } // called within the threadLoop to obtain timestamp from the HAL. virtual status_t threadloop_getHalTimestamp_l( ExtendedTimestamp *timestamp __unused) const { return INVALID_OPERATION; } virtual void dumpInternals_l(int fd __unused, const Vector& args __unused) { } virtual void dumpTracks_l(int fd __unused, const Vector& args __unused) { } friend class AudioFlinger; // for mEffectChains const type_t mType; // Used by parameters, config events, addTrack_l, exit Condition mWaitWorkCV; const sp mAudioFlinger; // updated by PlaybackThread::readOutputParameters_l() or // RecordThread::readInputParameters_l() uint32_t mSampleRate; size_t mFrameCount; // output HAL, direct output, record audio_channel_mask_t mChannelMask; uint32_t mChannelCount; size_t mFrameSize; // not HAL frame size, this is for output sink (to pipe to fast mixer) audio_format_t mFormat; // Source format for Recording and // Sink format for Playback. // Sink format may be different than // HAL format if Fastmixer is used. audio_format_t mHALFormat; size_t mBufferSize; // HAL buffer size for read() or write() Vector< sp > mConfigEvents; Vector< sp > mPendingConfigEvents; // events awaiting system ready // These fields are written and read by thread itself without lock or barrier, // and read by other threads without lock or barrier via standby(), outDevice() // and inDevice(). // Because of the absence of a lock or barrier, any other thread that reads // these fields must use the information in isolation, or be prepared to deal // with possibility that it might be inconsistent with other information. bool mStandby; // Whether thread is currently in standby. audio_devices_t mOutDevice; // output device audio_devices_t mInDevice; // input device audio_devices_t mPrevOutDevice; // previous output device audio_devices_t mPrevInDevice; // previous input device struct audio_patch mPatch; /** * @brief mDeviceId current device port unique identifier */ audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE; audio_source_t mAudioSource; const audio_io_handle_t mId; Vector< sp > mEffectChains; static const int kThreadNameLength = 16; // prctl(PR_SET_NAME) limit char mThreadName[kThreadNameLength]; // guaranteed NUL-terminated sp mPowerManager; sp mWakeLockToken; const sp mDeathRecipient; // list of suspended effects per session and per type. The first (outer) vector is // keyed by session ID, the second (inner) by type UUID timeLow field // Updated by updateSuspendedSessions_l() only. KeyedVector< audio_session_t, KeyedVector< int, sp > > mSuspendedSessions; // TODO: add comment and adjust size as needed static const size_t kLogSize = 4 * 1024; sp mNBLogWriter; bool mSystemReady; ExtendedTimestamp mTimestamp; TimestampVerifier< // For timestamp statistics. int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier; audio_devices_t mTimestampCorrectedDevices = AUDIO_DEVICE_NONE; // ThreadLoop statistics per iteration. int64_t mLastIoBeginNs = -1; int64_t mLastIoEndNs = -1; // This should be read under ThreadBase lock (if not on the threadLoop thread). audio_utils::Statistics mIoJitterMs{0.995 /* alpha */}; audio_utils::Statistics mProcessTimeMs{0.995 /* alpha */}; audio_utils::Statistics mLatencyMs{0.995 /* alpha */}; // Save the last count when we delivered statistics to mediametrics. int64_t mLastRecordedTimestampVerifierN = 0; int64_t mLastRecordedTimeNs = 0; // BOOTTIME to include suspend. bool mIsMsdDevice = false; // A condition that must be evaluated by the thread loop has changed and // we must not wait for async write callback in the thread loop before evaluating it bool mSignalPending; #ifdef TEE_SINK NBAIO_Tee mTee; #endif // ActiveTracks is a sorted vector of track type T representing the // active tracks of threadLoop() to be considered by the locked prepare portion. // ActiveTracks should be accessed with the ThreadBase lock held. // // During processing and I/O, the threadLoop does not hold the lock; // hence it does not directly use ActiveTracks. Care should be taken // to hold local strong references or defer removal of tracks // if the threadLoop may still be accessing those tracks due to mix, etc. // // This class updates power information appropriately. // template class ActiveTracks { public: explicit ActiveTracks(SimpleLog *localLog = nullptr) : mActiveTracksGeneration(0) , mLastActiveTracksGeneration(0) , mLocalLog(localLog) { } ~ActiveTracks() { ALOGW_IF(!mActiveTracks.isEmpty(), "ActiveTracks should be empty in destructor"); } // returns the last track added (even though it may have been // subsequently removed from ActiveTracks). // // Used for DirectOutputThread to ensure a flush is called when transitioning // to a new track (even though it may be on the same session). // Used for OffloadThread to ensure that volume and mixer state is // taken from the latest track added. // // The latest track is saved with a weak pointer to prevent keeping an // otherwise useless track alive. Thus the function will return nullptr // if the latest track has subsequently been removed and destroyed. sp getLatest() { return mLatestActiveTrack.promote(); } // SortedVector methods ssize_t add(const sp &track); ssize_t remove(const sp &track); size_t size() const { return mActiveTracks.size(); } bool isEmpty() const { return mActiveTracks.isEmpty(); } ssize_t indexOf(const sp& item) { return mActiveTracks.indexOf(item); } sp operator[](size_t index) const { return mActiveTracks[index]; } typename SortedVector>::iterator begin() { return mActiveTracks.begin(); } typename SortedVector>::iterator end() { return mActiveTracks.end(); } // Due to Binder recursion optimization, clear() and updatePowerState() // cannot be called from a Binder thread because they may call back into // the original calling process (system server) for BatteryNotifier // (which requires a Java environment that may not be present). // Hence, call clear() and updatePowerState() only from the // ThreadBase thread. void clear(); // periodically called in the threadLoop() to update power state uids. void updatePowerState(sp thread, bool force = false); /** @return true if one or move active tracks was added or removed since the * last time this function was called or the vector was created. */ bool readAndClearHasChanged(); private: void logTrack(const char *funcName, const sp &track) const; SortedVector getWakeLockUids() { SortedVector wakeLockUids; for (const sp &track : mActiveTracks) { wakeLockUids.add(track->uid()); } return wakeLockUids; // moved by underlying SharedBuffer } std::map> mBatteryCounter; SortedVector> mActiveTracks; int mActiveTracksGeneration; int mLastActiveTracksGeneration; wp mLatestActiveTrack; // latest track added to ActiveTracks SimpleLog * const mLocalLog; // If the vector has changed since last call to readAndClearHasChanged bool mHasChanged = false; }; SimpleLog mLocalLog; private: void dumpBase_l(int fd, const Vector& args); void dumpEffectChains_l(int fd, const Vector& args); }; class VolumeInterface { public: virtual ~VolumeInterface() {} virtual void setMasterVolume(float value) = 0; virtual void setMasterMute(bool muted) = 0; virtual void setStreamVolume(audio_stream_type_t stream, float value) = 0; virtual void setStreamMute(audio_stream_type_t stream, bool muted) = 0; virtual float streamVolume(audio_stream_type_t stream) const = 0; }; // --- PlaybackThread --- class PlaybackThread : public ThreadBase, public StreamOutHalInterfaceCallback, public VolumeInterface { public: #include "PlaybackTracks.h" enum mixer_state { MIXER_IDLE, // no active tracks MIXER_TRACKS_ENABLED, // at least one active track, but no track has any data ready MIXER_TRACKS_READY, // at least one active track, and at least one track has data MIXER_DRAIN_TRACK, // drain currently playing track MIXER_DRAIN_ALL, // fully drain the hardware // standby mode does not have an enum value // suspend by audio policy manager is orthogonal to mixer state }; // retry count before removing active track in case of underrun on offloaded thread: // we need to make sure that AudioTrack client has enough time to send large buffers //FIXME may be more appropriate if expressed in time units. Need to revise how underrun is // handled for offloaded tracks static const int8_t kMaxTrackRetriesOffload = 20; static const int8_t kMaxTrackStartupRetriesOffload = 100; static const int8_t kMaxTrackStopRetriesOffload = 2; static constexpr uint32_t kMaxTracksPerUid = 40; static constexpr size_t kMaxTracks = 256; // Maximum delay (in nanoseconds) for upcoming buffers in suspend mode, otherwise // if delay is greater, the estimated time for timeLoopNextNs is reset. // This allows for catch-up to be done for small delays, while resetting the estimate // for initial conditions or large delays. static const nsecs_t kMaxNextBufferDelayNs = 100000000; PlaybackThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, type_t type, bool systemReady); virtual ~PlaybackThread(); // Thread virtuals virtual bool threadLoop(); // RefBase virtual void onFirstRef(); virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, audio_session_t sessionId); protected: // Code snippets that were lifted up out of threadLoop() virtual void threadLoop_mix() = 0; virtual void threadLoop_sleepTime() = 0; virtual ssize_t threadLoop_write(); virtual void threadLoop_drain(); virtual void threadLoop_standby(); virtual void threadLoop_exit(); virtual void threadLoop_removeTracks(const Vector< sp >& tracksToRemove); // prepareTracks_l reads and writes mActiveTracks, and returns // the pending set of tracks to remove via Vector 'tracksToRemove'. The caller // is responsible for clearing or destroying this Vector later on, when it // is safe to do so. That will drop the final ref count and destroy the tracks. virtual mixer_state prepareTracks_l(Vector< sp > *tracksToRemove) = 0; void removeTracks_l(const Vector< sp >& tracksToRemove); // StreamOutHalInterfaceCallback implementation virtual void onWriteReady(); virtual void onDrainReady(); virtual void onError(); void resetWriteBlocked(uint32_t sequence); void resetDraining(uint32_t sequence); virtual bool waitingAsyncCallback(); virtual bool waitingAsyncCallback_l(); virtual bool shouldStandby_l(); virtual void onAddNewTrack_l(); void onAsyncError(); // error reported by AsyncCallbackThread // ThreadBase virtuals virtual void preExit(); virtual bool keepWakeLock() const { return true; } virtual void acquireWakeLock_l() { ThreadBase::acquireWakeLock_l(); mActiveTracks.updatePowerState(this, true /* force */); } void dumpInternals_l(int fd, const Vector& args) override; void dumpTracks_l(int fd, const Vector& args) override; public: virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; } // return estimated latency in milliseconds, as reported by HAL uint32_t latency() const; // same, but lock must already be held uint32_t latency_l() const; // VolumeInterface virtual void setMasterVolume(float value); virtual void setMasterBalance(float balance); virtual void setMasterMute(bool muted); virtual void setStreamVolume(audio_stream_type_t stream, float value); virtual void setStreamMute(audio_stream_type_t stream, bool muted); virtual float streamVolume(audio_stream_type_t stream) const; void setVolumeForOutput_l(float left, float right) const; sp createTrack_l( const sp& client, audio_stream_type_t streamType, const audio_attributes_t& attr, uint32_t *sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, size_t *pNotificationFrameCount, uint32_t notificationsPerBuffer, float speed, const sp& sharedBuffer, audio_session_t sessionId, audio_output_flags_t *flags, pid_t creatorPid, pid_t tid, uid_t uid, status_t *status /*non-NULL*/, audio_port_handle_t portId); AudioStreamOut* getOutput() const; AudioStreamOut* clearOutput(); virtual sp stream() const; // a very large number of suspend() will eventually wraparound, but unlikely void suspend() { (void) android_atomic_inc(&mSuspended); } void restore() { // if restore() is done without suspend(), get back into // range so that the next suspend() will operate correctly if (android_atomic_dec(&mSuspended) <= 0) { android_atomic_release_store(0, &mSuspended); } } bool isSuspended() const { return android_atomic_acquire_load(&mSuspended) > 0; } virtual String8 getParameters(const String8& keys); virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE); status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames); // Consider also removing and passing an explicit mMainBuffer initialization // parameter to AF::PlaybackThread::Track::Track(). effect_buffer_t *sinkBuffer() const { return reinterpret_cast(mSinkBuffer); }; virtual void detachAuxEffect_l(int effectId); status_t attachAuxEffect(const sp& track, int EffectId); status_t attachAuxEffect_l(const sp& track, int EffectId); virtual status_t addEffectChain_l(const sp& chain); virtual size_t removeEffectChain_l(const sp& chain); uint32_t hasAudioSession_l(audio_session_t sessionId) const override { return ThreadBase::hasAudioSession_l(sessionId, mTracks); } virtual uint32_t getStrategyForSession_l(audio_session_t sessionId); virtual status_t setSyncEvent(const sp& event); virtual bool isValidSyncEvent(const sp& event) const; // called with AudioFlinger lock held bool invalidateTracks_l(audio_stream_type_t streamType); virtual void invalidateTracks(audio_stream_type_t streamType); virtual size_t frameCount() const { return mNormalFrameCount; } status_t getTimestamp_l(AudioTimestamp& timestamp); void addPatchTrack(const sp& track); void deletePatchTrack(const sp& track); virtual void toAudioPortConfig(struct audio_port_config *config); // Return the asynchronous signal wait time. virtual int64_t computeWaitTimeNs_l() const { return INT64_MAX; } virtual bool isOutput() const override { return true; } // returns true if the track is allowed to be added to the thread. virtual bool isTrackAllowed_l( audio_channel_mask_t channelMask __unused, audio_format_t format __unused, audio_session_t sessionId __unused, uid_t uid) const { return trackCountForUid_l(uid) < PlaybackThread::kMaxTracksPerUid && mTracks.size() < PlaybackThread::kMaxTracks; } bool isTimestampCorrectionEnabled() const override { const audio_devices_t device = mOutDevice & mTimestampCorrectedDevices; return audio_is_output_devices(device) && popcount(device) > 0; } protected: // updated by readOutputParameters_l() size_t mNormalFrameCount; // normal mixer and effects bool mThreadThrottle; // throttle the thread processing uint32_t mThreadThrottleTimeMs; // throttle time for MIXER threads uint32_t mThreadThrottleEndMs; // notify once per throttling uint32_t mHalfBufferMs; // half the buffer size in milliseconds void* mSinkBuffer; // frame size aligned sink buffer // TODO: // Rearrange the buffer info into a struct/class with // clear, copy, construction, destruction methods. // // mSinkBuffer also has associated with it: // // mSinkBufferSize: Sink Buffer Size // mFormat: Sink Buffer Format // Mixer Buffer (mMixerBuffer*) // // In the case of floating point or multichannel data, which is not in the // sink format, it is required to accumulate in a higher precision or greater channel count // buffer before downmixing or data conversion to the sink buffer. // Set to "true" to enable the Mixer Buffer otherwise mixer output goes to sink buffer. bool mMixerBufferEnabled; // Storage, 32 byte aligned (may make this alignment a requirement later). // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. void* mMixerBuffer; // Size of mMixerBuffer in bytes: mNormalFrameCount * #channels * sampsize. size_t mMixerBufferSize; // The audio format of mMixerBuffer. Set to AUDIO_FORMAT_PCM_(FLOAT|16_BIT) only. audio_format_t mMixerBufferFormat; // An internal flag set to true by MixerThread::prepareTracks_l() // when mMixerBuffer contains valid data after mixing. bool mMixerBufferValid; // Effects Buffer (mEffectsBuffer*) // // In the case of effects data, which is not in the sink format, // it is required to accumulate in a different buffer before data conversion // to the sink buffer. // Set to "true" to enable the Effects Buffer otherwise effects output goes to sink buffer. bool mEffectBufferEnabled; // Storage, 32 byte aligned (may make this alignment a requirement later). // Due to constraints on mNormalFrameCount, the buffer size is a multiple of 16 frames. void* mEffectBuffer; // Size of mEffectsBuffer in bytes: mNormalFrameCount * #channels * sampsize. size_t mEffectBufferSize; // The audio format of mEffectsBuffer. Set to AUDIO_FORMAT_PCM_16_BIT only. audio_format_t mEffectBufferFormat; // An internal flag set to true by MixerThread::prepareTracks_l() // when mEffectsBuffer contains valid data after mixing. // // When this is set, all mixer data is routed into the effects buffer // for any processing (including output processing). bool mEffectBufferValid; // suspend count, > 0 means suspended. While suspended, the thread continues to pull from // tracks and mix, but doesn't write to HAL. A2DP and SCO HAL implementations can't handle // concurrent use of both of them, so Audio Policy Service suspends one of the threads to // workaround that restriction. // 'volatile' means accessed via atomic operations and no lock. volatile int32_t mSuspended; int64_t mBytesWritten; int64_t mFramesWritten; // not reset on standby int64_t mSuspendedFrames; // not reset on standby // mHapticChannelMask and mHapticChannelCount will only be valid when the thread support // haptic playback. audio_channel_mask_t mHapticChannelMask = AUDIO_CHANNEL_NONE; uint32_t mHapticChannelCount = 0; private: // mMasterMute is in both PlaybackThread and in AudioFlinger. When a // PlaybackThread needs to find out if master-muted, it checks it's local // copy rather than the one in AudioFlinger. This optimization saves a lock. bool mMasterMute; void setMasterMute_l(bool muted) { mMasterMute = muted; } protected: ActiveTracks mActiveTracks; // Time to sleep between cycles when: virtual uint32_t activeSleepTimeUs() const; // mixer state MIXER_TRACKS_ENABLED virtual uint32_t idleSleepTimeUs() const = 0; // mixer state MIXER_IDLE virtual uint32_t suspendSleepTimeUs() const = 0; // audio policy manager suspended us // No sleep when mixer state == MIXER_TRACKS_READY; relies on audio HAL stream->write() // No sleep in standby mode; waits on a condition // Code snippets that are temporarily lifted up out of threadLoop() until the merge void checkSilentMode_l(); // Non-trivial for DUPLICATING only virtual void saveOutputTracks() { } virtual void clearOutputTracks() { } // Cache various calculated values, at threadLoop() entry and after a parameter change virtual void cacheParameters_l(); virtual uint32_t correctLatency_l(uint32_t latency) const; virtual status_t createAudioPatch_l(const struct audio_patch *patch, audio_patch_handle_t *handle); virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); bool usesHwAvSync() const { return (mType == DIRECT) && (mOutput != NULL) && mHwSupportsPause && (mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC); } uint32_t trackCountForUid_l(uid_t uid) const; private: friend class AudioFlinger; // for numerous DISALLOW_COPY_AND_ASSIGN(PlaybackThread); status_t addTrack_l(const sp& track); bool destroyTrack_l(const sp& track); void removeTrack_l(const sp& track); void readOutputParameters_l(); void updateMetadata_l() final; virtual void sendMetadataToBackend_l(const StreamOutHalInterface::SourceMetadata& metadata); // The Tracks class manages tracks added and removed from the Thread. template class Tracks { public: Tracks(bool saveDeletedTrackIds) : mSaveDeletedTrackIds(saveDeletedTrackIds) { } // SortedVector methods ssize_t add(const sp &track) { const ssize_t index = mTracks.add(track); LOG_ALWAYS_FATAL_IF(index < 0, "cannot add track"); return index; } ssize_t remove(const sp &track); size_t size() const { return mTracks.size(); } bool isEmpty() const { return mTracks.isEmpty(); } ssize_t indexOf(const sp &item) { return mTracks.indexOf(item); } sp operator[](size_t index) const { return mTracks[index]; } typename SortedVector>::iterator begin() { return mTracks.begin(); } typename SortedVector>::iterator end() { return mTracks.end(); } size_t processDeletedTrackIds(std::function f) { for (const int trackId : mDeletedTrackIds) { f(trackId); } return mDeletedTrackIds.size(); } void clearDeletedTrackIds() { mDeletedTrackIds.clear(); } private: // Tracks pending deletion for MIXER type threads const bool mSaveDeletedTrackIds; // true to enable tracking std::set mDeletedTrackIds; SortedVector> mTracks; // wrapped SortedVector. }; Tracks mTracks; stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; AudioStreamOut *mOutput; float mMasterVolume; std::atomic mMasterBalance{}; audio_utils::Balance mBalance; int mNumWrites; int mNumDelayedWrites; bool mInWrite; // FIXME rename these former local variables of threadLoop to standard "m" names nsecs_t mStandbyTimeNs; size_t mSinkBufferSize; // cached copies of activeSleepTimeUs() and idleSleepTimeUs() made by cacheParameters_l() uint32_t mActiveSleepTimeUs; uint32_t mIdleSleepTimeUs; uint32_t mSleepTimeUs; // mixer status returned by prepareTracks_l() mixer_state mMixerStatus; // current cycle // previous cycle when in prepareTracks_l() mixer_state mMixerStatusIgnoringFastTracks; // FIXME or a separate ready state per track // FIXME move these declarations into the specific sub-class that needs them // MIXER only uint32_t sleepTimeShift; // same as AudioFlinger::mStandbyTimeInNsecs except for DIRECT which uses a shorter value nsecs_t mStandbyDelayNs; // MIXER only nsecs_t maxPeriod; // DUPLICATING only uint32_t writeFrames; size_t mBytesRemaining; size_t mCurrentWriteLength; bool mUseAsyncWrite; // mWriteAckSequence contains current write sequence on bits 31-1. The write sequence is // incremented each time a write(), a flush() or a standby() occurs. // Bit 0 is set when a write blocks and indicates a callback is expected. // Bit 0 is reset by the async callback thread calling resetWriteBlocked(). Out of sequence // callbacks are ignored. uint32_t mWriteAckSequence; // mDrainSequence contains current drain sequence on bits 31-1. The drain sequence is // incremented each time a drain is requested or a flush() or standby() occurs. // Bit 0 is set when the drain() command is called at the HAL and indicates a callback is // expected. // Bit 0 is reset by the async callback thread calling resetDraining(). Out of sequence // callbacks are ignored. uint32_t mDrainSequence; sp mCallbackThread; private: // The HAL output sink is treated as non-blocking, but current implementation is blocking sp mOutputSink; // If a fast mixer is present, the blocking pipe sink, otherwise clear sp mPipeSink; // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink sp mNormalSink; uint32_t mScreenState; // cached copy of gScreenState // TODO: add comment and adjust size as needed static const size_t kFastMixerLogSize = 8 * 1024; sp mFastMixerNBLogWriter; // Downstream patch latency, available if mDownstreamLatencyStatMs.getN() > 0. audio_utils::Statistics mDownstreamLatencyStatMs{0.999}; public: virtual bool hasFastMixer() const = 0; virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex __unused) const { FastTrackUnderruns dummy; return dummy; } protected: // accessed by both binder threads and within threadLoop(), lock on mutex needed unsigned mFastTrackAvailMask; // bit i set if fast track [i] is available bool mHwSupportsPause; bool mHwPaused; bool mFlushPending; // volumes last sent to audio HAL with stream->setVolume() float mLeftVolFloat; float mRightVolFloat; }; class MixerThread : public PlaybackThread { public: MixerThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type = MIXER); virtual ~MixerThread(); // Thread virtuals virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status); virtual bool isTrackAllowed_l( audio_channel_mask_t channelMask, audio_format_t format, audio_session_t sessionId, uid_t uid) const override; protected: virtual mixer_state prepareTracks_l(Vector< sp > *tracksToRemove); virtual uint32_t idleSleepTimeUs() const; virtual uint32_t suspendSleepTimeUs() const; virtual void cacheParameters_l(); virtual void acquireWakeLock_l() { PlaybackThread::acquireWakeLock_l(); if (hasFastMixer()) { mFastMixer->setBoottimeOffset( mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME]); } } void dumpInternals_l(int fd, const Vector& args) override; // threadLoop snippets virtual ssize_t threadLoop_write(); virtual void threadLoop_standby(); virtual void threadLoop_mix(); virtual void threadLoop_sleepTime(); virtual uint32_t correctLatency_l(uint32_t latency) const; virtual status_t createAudioPatch_l(const struct audio_patch *patch, audio_patch_handle_t *handle); virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); AudioMixer* mAudioMixer; // normal mixer private: // one-time initialization, no locks required sp mFastMixer; // non-0 if there is also a fast mixer sp mAudioWatchdog; // non-0 if there is an audio watchdog thread // contents are not guaranteed to be consistent, no locks required FastMixerDumpState mFastMixerDumpState; #ifdef STATE_QUEUE_DUMP StateQueueObserverDump mStateQueueObserverDump; StateQueueMutatorDump mStateQueueMutatorDump; #endif AudioWatchdogDump mAudioWatchdogDump; // accessible only within the threadLoop(), no locks required // mFastMixer->sq() // for mutating and pushing state int32_t mFastMixerFutex; // for cold idle std::atomic_bool mMasterMono; public: virtual bool hasFastMixer() const { return mFastMixer != 0; } virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const { ALOG_ASSERT(fastIndex < FastMixerState::sMaxFastTracks); return mFastMixerDumpState.mTracks[fastIndex].mUnderruns; } status_t threadloop_getHalTimestamp_l( ExtendedTimestamp *timestamp) const override { if (mNormalSink.get() != nullptr) { return mNormalSink->getTimestamp(*timestamp); } return INVALID_OPERATION; } protected: virtual void setMasterMono_l(bool mono) { mMasterMono.store(mono); if (mFastMixer != nullptr) { /* hasFastMixer() */ mFastMixer->setMasterMono(mMasterMono); } } // the FastMixer performs mono blend if it exists. // Blending with limiter is not idempotent, // and blending without limiter is idempotent but inefficient to do twice. virtual bool requireMonoBlend() { return mMasterMono.load() && !hasFastMixer(); } void setMasterBalance(float balance) override { mMasterBalance.store(balance); if (hasFastMixer()) { mFastMixer->setMasterBalance(balance); } } }; class DirectOutputThread : public PlaybackThread { public: DirectOutputThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) : DirectOutputThread(audioFlinger, output, id, device, DIRECT, systemReady) { } virtual ~DirectOutputThread(); status_t selectPresentation(int presentationId, int programId); // Thread virtuals virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status); virtual void flushHw_l(); void setMasterBalance(float balance) override; protected: virtual uint32_t activeSleepTimeUs() const; virtual uint32_t idleSleepTimeUs() const; virtual uint32_t suspendSleepTimeUs() const; virtual void cacheParameters_l(); void dumpInternals_l(int fd, const Vector& args) override; // threadLoop snippets virtual mixer_state prepareTracks_l(Vector< sp > *tracksToRemove); virtual void threadLoop_mix(); virtual void threadLoop_sleepTime(); virtual void threadLoop_exit(); virtual bool shouldStandby_l(); virtual void onAddNewTrack_l(); bool mVolumeShaperActive = false; DirectOutputThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, ThreadBase::type_t type, bool systemReady); void processVolume_l(Track *track, bool lastTrack); // prepareTracks_l() tells threadLoop_mix() the name of the single active track sp mActiveTrack; wp mPreviousTrack; // used to detect track switch // This must be initialized for initial condition of mMasterBalance = 0 (disabled). float mMasterBalanceLeft = 1.f; float mMasterBalanceRight = 1.f; public: virtual bool hasFastMixer() const { return false; } virtual int64_t computeWaitTimeNs_l() const override; status_t threadloop_getHalTimestamp_l(ExtendedTimestamp *timestamp) const override { // For DIRECT and OFFLOAD threads, query the output sink directly. if (mOutput != nullptr) { uint64_t uposition64; struct timespec time; if (mOutput->getPresentationPosition( &uposition64, &time) == OK) { timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL] = (int64_t)uposition64; timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = audio_utils_ns_from_timespec(&time); return NO_ERROR; } } return INVALID_OPERATION; } }; class OffloadThread : public DirectOutputThread { public: OffloadThread(const sp& audioFlinger, AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady); virtual ~OffloadThread() {}; virtual void flushHw_l(); protected: // threadLoop snippets virtual mixer_state prepareTracks_l(Vector< sp > *tracksToRemove); virtual void threadLoop_exit(); virtual bool waitingAsyncCallback(); virtual bool waitingAsyncCallback_l(); virtual void invalidateTracks(audio_stream_type_t streamType); virtual bool keepWakeLock() const { return (mKeepWakeLock || (mDrainSequence & 1)); } private: size_t mPausedWriteLength; // length in bytes of write interrupted by pause size_t mPausedBytesRemaining; // bytes still waiting in mixbuffer after resume bool mKeepWakeLock; // keep wake lock while waiting for write callback uint64_t mOffloadUnderrunPosition; // Current frame position for offloaded playback // used and valid only during underrun. ~0 if // no underrun has occurred during playback and // is not reset on standby. }; class AsyncCallbackThread : public Thread { public: explicit AsyncCallbackThread(const wp& playbackThread); virtual ~AsyncCallbackThread(); // Thread virtuals virtual bool threadLoop(); // RefBase virtual void onFirstRef(); void exit(); void setWriteBlocked(uint32_t sequence); void resetWriteBlocked(); void setDraining(uint32_t sequence); void resetDraining(); void setAsyncError(); private: const wp mPlaybackThread; // mWriteAckSequence corresponds to the last write sequence passed by the offload thread via // setWriteBlocked(). The sequence is shifted one bit to the left and the lsb is used // to indicate that the callback has been received via resetWriteBlocked() uint32_t mWriteAckSequence; // mDrainSequence corresponds to the last drain sequence passed by the offload thread via // setDraining(). The sequence is shifted one bit to the left and the lsb is used // to indicate that the callback has been received via resetDraining() uint32_t mDrainSequence; Condition mWaitWorkCV; Mutex mLock; bool mAsyncError; }; class DuplicatingThread : public MixerThread { public: DuplicatingThread(const sp& audioFlinger, MixerThread* mainThread, audio_io_handle_t id, bool systemReady); virtual ~DuplicatingThread(); // Thread virtuals void addOutputTrack(MixerThread* thread); void removeOutputTrack(MixerThread* thread); uint32_t waitTimeMs() const { return mWaitTimeMs; } void sendMetadataToBackend_l( const StreamOutHalInterface::SourceMetadata& metadata) override; protected: virtual uint32_t activeSleepTimeUs() const; void dumpInternals_l(int fd, const Vector& args) override; private: bool outputsReady(const SortedVector< sp > &outputTracks); protected: // threadLoop snippets virtual void threadLoop_mix(); virtual void threadLoop_sleepTime(); virtual ssize_t threadLoop_write(); virtual void threadLoop_standby(); virtual void cacheParameters_l(); private: // called from threadLoop, addOutputTrack, removeOutputTrack virtual void updateWaitTime_l(); protected: virtual void saveOutputTracks(); virtual void clearOutputTracks(); private: uint32_t mWaitTimeMs; SortedVector < sp > outputTracks; SortedVector < sp > mOutputTracks; public: virtual bool hasFastMixer() const { return false; } status_t threadloop_getHalTimestamp_l( ExtendedTimestamp *timestamp) const override { if (mOutputTracks.size() > 0) { // forward the first OutputTrack's kernel information for timestamp. const ExtendedTimestamp trackTimestamp = mOutputTracks[0]->getClientProxyTimestamp(); if (trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] > 0) { timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL] = trackTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; return OK; // discard server timestamp - that's ignored. } } return INVALID_OPERATION; } }; // record thread class RecordThread : public ThreadBase { public: class RecordTrack; /* The ResamplerBufferProvider is used to retrieve recorded input data from the * RecordThread. It maintains local state on the relative position of the read * position of the RecordTrack compared with the RecordThread. */ class ResamplerBufferProvider : public AudioBufferProvider { public: explicit ResamplerBufferProvider(RecordTrack* recordTrack) : mRecordTrack(recordTrack), mRsmpInUnrel(0), mRsmpInFront(0) { } virtual ~ResamplerBufferProvider() { } // called to set the ResamplerBufferProvider to head of the RecordThread data buffer, // skipping any previous data read from the hal. virtual void reset(); /* Synchronizes RecordTrack position with the RecordThread. * Calculates available frames and handle overruns if the RecordThread * has advanced faster than the ResamplerBufferProvider has retrieved data. * TODO: why not do this for every getNextBuffer? * * Parameters * framesAvailable: pointer to optional output size_t to store record track * frames available. * hasOverrun: pointer to optional boolean, returns true if track has overrun. */ virtual void sync(size_t *framesAvailable = NULL, bool *hasOverrun = NULL); // AudioBufferProvider interface virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); private: RecordTrack * const mRecordTrack; size_t mRsmpInUnrel; // unreleased frames remaining from // most recent getNextBuffer // for debug only int32_t mRsmpInFront; // next available frame // rolling counter that is never cleared }; #include "RecordTracks.h" RecordThread(const sp& audioFlinger, AudioStreamIn *input, audio_io_handle_t id, audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady ); virtual ~RecordThread(); // no addTrack_l ? void destroyTrack_l(const sp& track); void removeTrack_l(const sp& track); // Thread virtuals virtual bool threadLoop(); virtual void preExit(); // RefBase virtual void onFirstRef(); virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; } virtual sp readOnlyHeap() const { return mReadOnlyHeap; } virtual sp pipeMemory() const { return mPipeMemory; } sp createRecordTrack_l( const sp& client, const audio_attributes_t& attr, uint32_t *pSampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t *pFrameCount, audio_session_t sessionId, size_t *pNotificationFrameCount, pid_t creatorPid, uid_t uid, audio_input_flags_t *flags, pid_t tid, status_t *status /*non-NULL*/, audio_port_handle_t portId); status_t start(RecordTrack* recordTrack, AudioSystem::sync_event_t event, audio_session_t triggerSession); // ask the thread to stop the specified track, and // return true if the caller should then do it's part of the stopping process bool stop(RecordTrack* recordTrack); AudioStreamIn* clearInput(); virtual sp stream() const; virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status); virtual void cacheParameters_l() {} virtual String8 getParameters(const String8& keys); virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE); virtual status_t createAudioPatch_l(const struct audio_patch *patch, audio_patch_handle_t *handle); virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); void addPatchTrack(const sp& record); void deletePatchTrack(const sp& record); void readInputParameters_l(); virtual uint32_t getInputFramesLost(); virtual status_t addEffectChain_l(const sp& chain); virtual size_t removeEffectChain_l(const sp& chain); uint32_t hasAudioSession_l(audio_session_t sessionId) const override { return ThreadBase::hasAudioSession_l(sessionId, mTracks); } // Return the set of unique session IDs across all tracks. // The keys are the session IDs, and the associated values are meaningless. // FIXME replace by Set [and implement Bag/Multiset for other uses]. KeyedVector sessionIds() const; virtual status_t setSyncEvent(const sp& event); virtual bool isValidSyncEvent(const sp& event) const; static void syncStartEventCallback(const wp& event); virtual size_t frameCount() const { return mFrameCount; } bool hasFastCapture() const { return mFastCapture != 0; } virtual void toAudioPortConfig(struct audio_port_config *config); virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, audio_session_t sessionId); virtual void acquireWakeLock_l() { ThreadBase::acquireWakeLock_l(); mActiveTracks.updatePowerState(this, true /* force */); } virtual bool isOutput() const override { return false; } void checkBtNrec(); // Sets the UID records silence void setRecordSilenced(uid_t uid, bool silenced); status_t getActiveMicrophones(std::vector* activeMicrophones); status_t setPreferredMicrophoneDirection(audio_microphone_direction_t direction); status_t setPreferredMicrophoneFieldDimension(float zoom); void updateMetadata_l() override; bool fastTrackAvailable() const { return mFastTrackAvail; } bool isTimestampCorrectionEnabled() const override { // checks popcount for exactly one device. return audio_is_input_device( mInDevice & mTimestampCorrectedDevices); } protected: void dumpInternals_l(int fd, const Vector& args) override; void dumpTracks_l(int fd, const Vector& args) override; private: // Enter standby if not already in standby, and set mStandby flag void standbyIfNotAlreadyInStandby(); // Call the HAL standby method unconditionally, and don't change mStandby flag void inputStandBy(); void checkBtNrec_l(); AudioStreamIn *mInput; SortedVector < sp > mTracks; // mActiveTracks has dual roles: it indicates the current active track(s), and // is used together with mStartStopCond to indicate start()/stop() progress ActiveTracks mActiveTracks; Condition mStartStopCond; // resampler converts input at HAL Hz to output at AudioRecord client Hz void *mRsmpInBuffer; // size = mRsmpInFramesOA size_t mRsmpInFrames; // size of resampler input in frames size_t mRsmpInFramesP2;// size rounded up to a power-of-2 size_t mRsmpInFramesOA;// mRsmpInFramesP2 + over-allocation // rolling index that is never cleared int32_t mRsmpInRear; // last filled frame + 1 // For dumpsys const sp mReadOnlyHeap; // one-time initialization, no locks required sp mFastCapture; // non-0 if there is also // a fast capture // FIXME audio watchdog thread // contents are not guaranteed to be consistent, no locks required FastCaptureDumpState mFastCaptureDumpState; #ifdef STATE_QUEUE_DUMP // FIXME StateQueue observer and mutator dump fields #endif // FIXME audio watchdog dump // accessible only within the threadLoop(), no locks required // mFastCapture->sq() // for mutating and pushing state int32_t mFastCaptureFutex; // for cold idle // The HAL input source is treated as non-blocking, // but current implementation is blocking sp mInputSource; // The source for the normal capture thread to read from: mInputSource or mPipeSource sp mNormalSource; // If a fast capture is present, the non-blocking pipe sink written to by fast capture, // otherwise clear sp mPipeSink; // If a fast capture is present, the non-blocking pipe source read by normal thread, // otherwise clear sp mPipeSource; // Depth of pipe from fast capture to normal thread and fast clients, always power of 2 size_t mPipeFramesP2; // If a fast capture is present, the Pipe as IMemory, otherwise clear sp mPipeMemory; // TODO: add comment and adjust size as needed static const size_t kFastCaptureLogSize = 4 * 1024; sp mFastCaptureNBLogWriter; bool mFastTrackAvail; // true if fast track available // common state to all record threads std::atomic_bool mBtNrecSuspended; int64_t mFramesRead = 0; // continuous running counter. }; class MmapThread : public ThreadBase { public: #include "MmapTracks.h" MmapThread(const sp& audioFlinger, audio_io_handle_t id, AudioHwDevice *hwDev, sp stream, audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); virtual ~MmapThread(); virtual void configure(const audio_attributes_t *attr, audio_stream_type_t streamType, audio_session_t sessionId, const sp& callback, audio_port_handle_t deviceId, audio_port_handle_t portId); void disconnect(); // MmapStreamInterface status_t createMmapBuffer(int32_t minSizeFrames, struct audio_mmap_buffer_info *info); status_t getMmapPosition(struct audio_mmap_position *position); status_t start(const AudioClient& client, audio_port_handle_t *handle); status_t stop(audio_port_handle_t handle); status_t standby(); // RefBase virtual void onFirstRef(); // Thread virtuals virtual bool threadLoop(); virtual void threadLoop_exit(); virtual void threadLoop_standby(); virtual bool shouldStandby_l() { return false; } virtual status_t exitStandby(); virtual status_t initCheck() const { return (mHalStream == 0) ? NO_INIT : NO_ERROR; } virtual size_t frameCount() const { return mFrameCount; } virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status); virtual String8 getParameters(const String8& keys); virtual void ioConfigChanged(audio_io_config_event event, pid_t pid = 0, audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE); void readHalParameters_l(); virtual void cacheParameters_l() {} virtual status_t createAudioPatch_l(const struct audio_patch *patch, audio_patch_handle_t *handle); virtual status_t releaseAudioPatch_l(const audio_patch_handle_t handle); virtual void toAudioPortConfig(struct audio_port_config *config); virtual sp stream() const { return mHalStream; } virtual status_t addEffectChain_l(const sp& chain); virtual size_t removeEffectChain_l(const sp& chain); virtual status_t checkEffectCompatibility_l(const effect_descriptor_t *desc, audio_session_t sessionId); uint32_t hasAudioSession_l(audio_session_t sessionId) const override { // Note: using mActiveTracks as no mTracks here. return ThreadBase::hasAudioSession_l(sessionId, mActiveTracks); } virtual status_t setSyncEvent(const sp& event); virtual bool isValidSyncEvent(const sp& event) const; virtual void checkSilentMode_l() {} virtual void processVolume_l() {} void checkInvalidTracks_l(); virtual audio_stream_type_t streamType() { return AUDIO_STREAM_DEFAULT; } virtual void invalidateTracks(audio_stream_type_t streamType __unused) {} // Sets the UID records silence virtual void setRecordSilenced(uid_t uid __unused, bool silenced __unused) {} protected: void dumpInternals_l(int fd, const Vector& args) override; void dumpTracks_l(int fd, const Vector& args) override; audio_attributes_t mAttr; audio_session_t mSessionId; audio_port_handle_t mPortId; wp mCallback; sp mHalStream; sp mHalDevice; AudioHwDevice* const mAudioHwDev; ActiveTracks mActiveTracks; float mHalVolFloat; int32_t mNoCallbackWarningCount; static constexpr int32_t kMaxNoCallbackWarnings = 5; }; class MmapPlaybackThread : public MmapThread, public VolumeInterface { public: MmapPlaybackThread(const sp& audioFlinger, audio_io_handle_t id, AudioHwDevice *hwDev, AudioStreamOut *output, audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); virtual ~MmapPlaybackThread() {} virtual void configure(const audio_attributes_t *attr, audio_stream_type_t streamType, audio_session_t sessionId, const sp& callback, audio_port_handle_t deviceId, audio_port_handle_t portId); AudioStreamOut* clearOutput(); // VolumeInterface virtual void setMasterVolume(float value); virtual void setMasterMute(bool muted); virtual void setStreamVolume(audio_stream_type_t stream, float value); virtual void setStreamMute(audio_stream_type_t stream, bool muted); virtual float streamVolume(audio_stream_type_t stream) const; void setMasterMute_l(bool muted) { mMasterMute = muted; } virtual void invalidateTracks(audio_stream_type_t streamType); virtual audio_stream_type_t streamType() { return mStreamType; } virtual void checkSilentMode_l(); void processVolume_l() override; virtual bool isOutput() const override { return true; } void updateMetadata_l() override; virtual void toAudioPortConfig(struct audio_port_config *config); protected: void dumpInternals_l(int fd, const Vector& args) override; audio_stream_type_t mStreamType; float mMasterVolume; float mStreamVolume; bool mMasterMute; bool mStreamMute; AudioStreamOut* mOutput; }; class MmapCaptureThread : public MmapThread { public: MmapCaptureThread(const sp& audioFlinger, audio_io_handle_t id, AudioHwDevice *hwDev, AudioStreamIn *input, audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady); virtual ~MmapCaptureThread() {} AudioStreamIn* clearInput(); status_t exitStandby() override; virtual bool isOutput() const override { return false; } void updateMetadata_l() override; void processVolume_l() override; void setRecordSilenced(uid_t uid, bool silenced) override; virtual void toAudioPortConfig(struct audio_port_config *config); protected: AudioStreamIn* mInput; };