| /* |
| ** |
| ** 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 |
| |
| // Checks and monitors OP_PLAY_AUDIO |
| class OpPlayAudioMonitor : public RefBase { |
| public: |
| ~OpPlayAudioMonitor() override; |
| bool hasOpPlayAudio() const; |
| |
| static sp<OpPlayAudioMonitor> createIfNeeded( |
| const AttributionSourceState& attributionSource, |
| const audio_attributes_t& attr, int id, |
| audio_stream_type_t streamType); |
| |
| private: |
| OpPlayAudioMonitor(const AttributionSourceState& attributionSource, |
| audio_usage_t usage, int id); |
| void onFirstRef() override; |
| static void getPackagesForUid(uid_t uid, Vector<String16>& packages); |
| |
| AppOpsManager mAppOpsManager; |
| |
| class PlayAudioOpCallback : public BnAppOpsCallback { |
| public: |
| explicit PlayAudioOpCallback(const wp<OpPlayAudioMonitor>& monitor); |
| void opChanged(int32_t op, const String16& packageName) override; |
| |
| private: |
| const wp<OpPlayAudioMonitor> mMonitor; |
| }; |
| |
| sp<PlayAudioOpCallback> mOpCallback; |
| // called by PlayAudioOpCallback when OP_PLAY_AUDIO is updated in AppOp callback |
| void checkPlayAudioForUsage(); |
| |
| std::atomic_bool mHasOpPlayAudio; |
| const AttributionSourceState mAttributionSource; |
| const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t |
| const int mId; // for logging purposes only |
| }; |
| |
| // playback track |
| class Track : public TrackBase, public VolumeProvider { |
| public: |
| Track( PlaybackThread *thread, |
| const sp<Client>& 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 frameCount, |
| void *buffer, |
| size_t bufferSize, |
| const sp<IMemory>& sharedBuffer, |
| audio_session_t sessionId, |
| pid_t creatorPid, |
| const AttributionSourceState& attributionSource, |
| audio_output_flags_t flags, |
| track_type type, |
| audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE, |
| /** default behaviour is to start when there are as many frames |
| * ready as possible (aka. Buffer is full). */ |
| size_t frameCountToBeReady = SIZE_MAX, |
| float speed = 1.0f); |
| virtual ~Track(); |
| virtual status_t initCheck() const; |
| |
| void appendDumpHeader(String8& result); |
| void appendDump(String8& result, bool active); |
| virtual status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE, |
| audio_session_t triggerSession = AUDIO_SESSION_NONE); |
| virtual void stop(); |
| void pause(); |
| |
| void flush(); |
| void destroy(); |
| |
| virtual uint32_t sampleRate() const; |
| |
| audio_stream_type_t streamType() const { |
| return mStreamType; |
| } |
| bool isOffloaded() const |
| { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; } |
| bool isDirect() const override |
| { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; } |
| bool isOffloadedOrDirect() const { return (mFlags |
| & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
| | AUDIO_OUTPUT_FLAG_DIRECT)) != 0; } |
| bool isStatic() const { return mSharedBuffer.get() != nullptr; } |
| |
| status_t setParameters(const String8& keyValuePairs); |
| status_t selectPresentation(int presentationId, int programId); |
| status_t attachAuxEffect(int EffectId); |
| void setAuxBuffer(int EffectId, int32_t *buffer); |
| int32_t *auxBuffer() const { return mAuxBuffer; } |
| void setMainBuffer(effect_buffer_t *buffer) { mMainBuffer = buffer; } |
| effect_buffer_t *mainBuffer() const { return mMainBuffer; } |
| int auxEffectId() const { return mAuxEffectId; } |
| virtual status_t getTimestamp(AudioTimestamp& timestamp); |
| void signal(); |
| status_t getDualMonoMode(audio_dual_mono_mode_t* mode); |
| status_t setDualMonoMode(audio_dual_mono_mode_t mode); |
| status_t getAudioDescriptionMixLevel(float* leveldB); |
| status_t setAudioDescriptionMixLevel(float leveldB); |
| status_t getPlaybackRateParameters(audio_playback_rate_t* playbackRate); |
| status_t setPlaybackRateParameters(const audio_playback_rate_t& playbackRate); |
| |
| // implement FastMixerState::VolumeProvider interface |
| virtual gain_minifloat_packed_t getVolumeLR(); |
| |
| virtual status_t setSyncEvent(const sp<SyncEvent>& event); |
| |
| virtual bool isFastTrack() const { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; } |
| |
| double bufferLatencyMs() const override { |
| return isStatic() ? 0. : TrackBase::bufferLatencyMs(); |
| } |
| |
| // implement volume handling. |
| media::VolumeShaper::Status applyVolumeShaper( |
| const sp<media::VolumeShaper::Configuration>& configuration, |
| const sp<media::VolumeShaper::Operation>& operation); |
| sp<media::VolumeShaper::State> getVolumeShaperState(int id); |
| sp<media::VolumeHandler> getVolumeHandler() { return mVolumeHandler; } |
| /** Set the computed normalized final volume of the track. |
| * !masterMute * masterVolume * streamVolume * averageLRVolume */ |
| void setFinalVolume(float volume); |
| float getFinalVolume() const { return mFinalVolume; } |
| |
| using SourceMetadatas = std::vector<playback_track_metadata_v7_t>; |
| using MetadataInserter = std::back_insert_iterator<SourceMetadatas>; |
| /** Copy the track metadata in the provided iterator. Thread safe. */ |
| virtual void copyMetadataTo(MetadataInserter& backInserter) const; |
| |
| /** Return haptic playback of the track is enabled or not, used in mixer. */ |
| bool getHapticPlaybackEnabled() const { return mHapticPlaybackEnabled; } |
| /** Set haptic playback of the track is enabled or not, should be |
| * set after query or get callback from vibrator service */ |
| void setHapticPlaybackEnabled(bool hapticPlaybackEnabled) { |
| mHapticPlaybackEnabled = hapticPlaybackEnabled; |
| } |
| /** Return at what intensity to play haptics, used in mixer. */ |
| os::HapticScale getHapticIntensity() const { return mHapticIntensity; } |
| /** Set intensity of haptic playback, should be set after querying vibrator service. */ |
| void setHapticIntensity(os::HapticScale hapticIntensity) { |
| if (os::isValidHapticScale(hapticIntensity)) { |
| mHapticIntensity = hapticIntensity; |
| setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE); |
| } |
| } |
| sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; } |
| |
| void setTeePatches(TeePatches teePatches); |
| |
| void tallyUnderrunFrames(size_t frames) override { |
| if (isOut()) { // we expect this from output tracks only |
| mAudioTrackServerProxy->tallyUnderrunFrames(frames); |
| // Fetch absolute numbers from AudioTrackShared as it counts |
| // contiguous underruns as a one -- we want a consistent number. |
| // TODO: isolate this counting into a class. |
| mTrackMetrics.logUnderruns(mAudioTrackServerProxy->getUnderrunCount(), |
| mAudioTrackServerProxy->getUnderrunFrames()); |
| } |
| } |
| |
| audio_output_flags_t getOutputFlags() const { return mFlags; } |
| float getSpeed() const { return mSpeed; } |
| protected: |
| // for numerous |
| friend class PlaybackThread; |
| friend class MixerThread; |
| friend class DirectOutputThread; |
| friend class OffloadThread; |
| |
| DISALLOW_COPY_AND_ASSIGN(Track); |
| |
| // AudioBufferProvider interface |
| status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) override; |
| void releaseBuffer(AudioBufferProvider::Buffer* buffer) override; |
| |
| // ExtendedAudioBufferProvider interface |
| virtual size_t framesReady() const; |
| virtual int64_t framesReleased() const; |
| virtual void onTimestamp(const ExtendedTimestamp ×tamp); |
| |
| bool isPausing() const { return mState == PAUSING; } |
| bool isPaused() const { return mState == PAUSED; } |
| bool isResuming() const { return mState == RESUMING; } |
| bool isReady() const; |
| void setPaused() { mState = PAUSED; } |
| void reset(); |
| bool isFlushPending() const { return mFlushHwPending; } |
| void flushAck(); |
| bool isResumePending(); |
| void resumeAck(); |
| // For direct or offloaded tracks ensure that the pause state is acknowledged |
| // by the playback thread in case of an immediate flush. |
| bool isPausePending() const { return mPauseHwPending; } |
| void pauseAck(); |
| void updateTrackFrameInfo(int64_t trackFramesReleased, int64_t sinkFramesWritten, |
| uint32_t halSampleRate, const ExtendedTimestamp &timeStamp); |
| |
| sp<IMemory> sharedBuffer() const { return mSharedBuffer; } |
| |
| // presentationComplete checked by frames. (Mixed Tracks). |
| // framesWritten is cumulative, never reset, and is shared all tracks |
| // audioHalFrames is derived from output latency |
| bool presentationComplete(int64_t framesWritten, size_t audioHalFrames); |
| |
| // presentationComplete checked by time. (Direct Tracks). |
| bool presentationComplete(uint32_t latencyMs); |
| |
| void resetPresentationComplete() { |
| mPresentationCompleteFrames = 0; |
| mPresentationCompleteTimeNs = 0; |
| } |
| |
| // notifyPresentationComplete is called when presentationComplete() detects |
| // that the track is finished stopping. |
| void notifyPresentationComplete(); |
| |
| void signalClientFlag(int32_t flag); |
| |
| public: |
| void triggerEvents(AudioSystem::sync_event_t type); |
| virtual void invalidate(); |
| void disable(); |
| |
| int fastIndex() const { return mFastIndex; } |
| |
| bool isPlaybackRestricted() const { |
| // The monitor is only created for tracks that can be silenced. |
| return mOpPlayAudioMonitor ? !mOpPlayAudioMonitor->hasOpPlayAudio() : false; } |
| |
| protected: |
| |
| // FILLED state is used for suppressing volume ramp at begin of playing |
| enum {FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE}; |
| mutable uint8_t mFillingUpStatus; |
| int8_t mRetryCount; |
| |
| // see comment at AudioFlinger::PlaybackThread::Track::~Track for why this can't be const |
| sp<IMemory> mSharedBuffer; |
| |
| bool mResetDone; |
| const audio_stream_type_t mStreamType; |
| effect_buffer_t *mMainBuffer; |
| |
| int32_t *mAuxBuffer; |
| int mAuxEffectId; |
| bool mHasVolumeController; |
| |
| // access these three variables only when holding thread lock. |
| LinearMap<int64_t> mFrameMap; // track frame to server frame mapping |
| |
| ExtendedTimestamp mSinkTimestamp; |
| |
| sp<media::VolumeHandler> mVolumeHandler; // handles multiple VolumeShaper configs and operations |
| |
| sp<OpPlayAudioMonitor> mOpPlayAudioMonitor; |
| |
| bool mHapticPlaybackEnabled = false; // indicates haptic playback enabled or not |
| // intensity to play haptic data |
| os::HapticScale mHapticIntensity = os::HapticScale::MUTE; |
| class AudioVibrationController : public os::BnExternalVibrationController { |
| public: |
| explicit AudioVibrationController(Track* track) : mTrack(track) {} |
| binder::Status mute(/*out*/ bool *ret) override; |
| binder::Status unmute(/*out*/ bool *ret) override; |
| private: |
| Track* const mTrack; |
| }; |
| sp<AudioVibrationController> mAudioVibrationController; |
| sp<os::ExternalVibration> mExternalVibration; |
| |
| audio_dual_mono_mode_t mDualMonoMode = AUDIO_DUAL_MONO_MODE_OFF; |
| float mAudioDescriptionMixLevel = -std::numeric_limits<float>::infinity(); |
| audio_playback_rate_t mPlaybackRateParameters = AUDIO_PLAYBACK_RATE_INITIALIZER; |
| |
| private: |
| void interceptBuffer(const AudioBufferProvider::Buffer& buffer); |
| template <class F> |
| void forEachTeePatchTrack(F f) { |
| for (auto& tp : mTeePatches) { f(tp.patchTrack); } |
| }; |
| |
| size_t mPresentationCompleteFrames = 0; // (Used for Mixed tracks) |
| // The number of frames written to the |
| // audio HAL when this track is considered fully rendered. |
| // Zero means not monitoring. |
| int64_t mPresentationCompleteTimeNs = 0; // (Used for Direct tracks) |
| // The time when this track is considered fully rendered. |
| // Zero means not monitoring. |
| |
| // The following fields are only for fast tracks, and should be in a subclass |
| int mFastIndex; // index within FastMixerState::mFastTracks[]; |
| // either mFastIndex == -1 if not isFastTrack() |
| // or 0 < mFastIndex < FastMixerState::kMaxFast because |
| // index 0 is reserved for normal mixer's submix; |
| // index is allocated statically at track creation time |
| // but the slot is only used if track is active |
| FastTrackUnderruns mObservedUnderruns; // Most recently observed value of |
| // mFastMixerDumpState.mTracks[mFastIndex].mUnderruns |
| volatile float mCachedVolume; // combined master volume and stream type volume; |
| // 'volatile' means accessed without lock or |
| // barrier, but is read/written atomically |
| float mFinalVolume; // combine master volume, stream type volume and track volume |
| sp<AudioTrackServerProxy> mAudioTrackServerProxy; |
| bool mResumeToStopping; // track was paused in stopping state. |
| bool mFlushHwPending; // track requests for thread flush |
| bool mPauseHwPending = false; // direct/offload track request for thread pause |
| audio_output_flags_t mFlags; |
| TeePatches mTeePatches; |
| const float mSpeed; |
| }; // end of Track |
| |
| |
| // playback track, used by DuplicatingThread |
| class OutputTrack : public Track { |
| public: |
| |
| class Buffer : public AudioBufferProvider::Buffer { |
| public: |
| void *mBuffer; |
| }; |
| |
| OutputTrack(PlaybackThread *thread, |
| DuplicatingThread *sourceThread, |
| uint32_t sampleRate, |
| audio_format_t format, |
| audio_channel_mask_t channelMask, |
| size_t frameCount, |
| const AttributionSourceState& attributionSource); |
| virtual ~OutputTrack(); |
| |
| virtual status_t start(AudioSystem::sync_event_t event = |
| AudioSystem::SYNC_EVENT_NONE, |
| audio_session_t triggerSession = AUDIO_SESSION_NONE); |
| virtual void stop(); |
| ssize_t write(void* data, uint32_t frames); |
| bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; } |
| bool isActive() const { return mActive; } |
| const wp<ThreadBase>& thread() const { return mThread; } |
| |
| void copyMetadataTo(MetadataInserter& backInserter) const override; |
| /** Set the metadatas of the upstream tracks. Thread safe. */ |
| void setMetadatas(const SourceMetadatas& metadatas); |
| /** returns client timestamp to the upstream duplicating thread. */ |
| ExtendedTimestamp getClientProxyTimestamp() const { |
| // server - kernel difference is not true latency when drained |
| // i.e. mServerProxy->isDrained(). |
| ExtendedTimestamp timestamp; |
| (void) mClientProxy->getTimestamp(×tamp); |
| // On success, the timestamp LOCATION_SERVER and LOCATION_KERNEL |
| // entries will be properly filled. If getTimestamp() |
| // is unsuccessful, then a default initialized timestamp |
| // (with mTimeNs[] filled with -1's) is returned. |
| return timestamp; |
| } |
| |
| private: |
| status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, |
| uint32_t waitTimeMs); |
| void clearBufferQueue(); |
| |
| void restartIfDisabled(); |
| |
| // Maximum number of pending buffers allocated by OutputTrack::write() |
| static const uint8_t kMaxOverFlowBuffers = 10; |
| |
| Vector < Buffer* > mBufferQueue; |
| AudioBufferProvider::Buffer mOutBuffer; |
| bool mActive; |
| DuplicatingThread* const mSourceThread; // for waitTimeMs() in write() |
| sp<AudioTrackClientProxy> mClientProxy; |
| |
| /** Attributes of the source tracks. |
| * |
| * This member must be accessed with mTrackMetadatasMutex taken. |
| * There is one writer (duplicating thread) and one reader (downstream mixer). |
| * |
| * That means that the duplicating thread can block the downstream mixer |
| * thread and vice versa for the time of the copy. |
| * If this becomes an issue, the metadata could be stored in an atomic raw pointer, |
| * and a exchange with nullptr and delete can be used. |
| * Alternatively a read-copy-update might be implemented. |
| */ |
| SourceMetadatas mTrackMetadatas; |
| /** Protects mTrackMetadatas against concurrent access. */ |
| mutable std::mutex mTrackMetadatasMutex; |
| }; // end of OutputTrack |
| |
| // playback track, used by PatchPanel |
| class PatchTrack : public Track, public PatchTrackBase { |
| public: |
| |
| PatchTrack(PlaybackThread *playbackThread, |
| audio_stream_type_t streamType, |
| uint32_t sampleRate, |
| audio_channel_mask_t channelMask, |
| audio_format_t format, |
| size_t frameCount, |
| void *buffer, |
| size_t bufferSize, |
| audio_output_flags_t flags, |
| const Timeout& timeout = {}, |
| size_t frameCountToBeReady = 1 /** Default behaviour is to start |
| * as soon as possible to have |
| * the lowest possible latency |
| * even if it might glitch. */); |
| virtual ~PatchTrack(); |
| |
| size_t framesReady() const override; |
| |
| virtual status_t start(AudioSystem::sync_event_t event = |
| AudioSystem::SYNC_EVENT_NONE, |
| audio_session_t triggerSession = AUDIO_SESSION_NONE); |
| |
| // AudioBufferProvider interface |
| virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer); |
| virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer); |
| |
| // PatchProxyBufferProvider interface |
| virtual status_t obtainBuffer(Proxy::Buffer* buffer, |
| const struct timespec *timeOut = NULL); |
| virtual void releaseBuffer(Proxy::Buffer* buffer); |
| |
| private: |
| void restartIfDisabled(); |
| }; // end of PatchTrack |
