services/audioflinger/PlaybackTracks.h

/*
**
** 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

#include <math.h>

// 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,
                                bool isSpatialized = false);
    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; }
            /** Return the maximum amplitude allowed for haptics data, used in mixer. */
            float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
            /** 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);
                }
            }
            /** Set maximum amplitude allowed for haptic data, should be set after querying
             *  vibrator service.
             */
            void    setHapticMaxAmplitude(float maxAmplitude) {
                mHapticMaxAmplitude = maxAmplitude;
            }
            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());
       }
    }

    static bool checkServerLatencySupported(
            audio_format_t format, audio_output_flags_t flags) {
        return audio_is_linear_pcm(format)
                && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == 0;
    }

    audio_output_flags_t getOutputFlags() const { return mFlags; }
    float getSpeed() const { return mSpeed; }
    bool isSpatialized() const override { return mIsSpatialized; }

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 &timestamp);

    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;
    // max amplitude allowed for haptic data
    float mHapticMaxAmplitude = NAN;
    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;
    const bool          mIsSpatialized;
};  // 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(&timestamp);
                            // 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