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Commit 8d31fd23 authored by Andy Hung's avatar Andy Hung
Browse files

AudioFlinger: Extract inner Track classes 

Finish removing inner Track classes from AudioFlinger.

Test: atest audiorecord_tests audiotrack_tests audiorouting_tests trackplayerbase_tests audiosystem_tests
Test: atest AudioTrackTest AudioRecordTest
Test: YouTube and Camera
Bug: 288339104
Bug: 288468076
Change-Id: Ib44c7eb854f9d447aa11223d76373b00c413a778
parent ca9be059

services/audioflinger/AudioFlinger.cpp

+3 −3
Original line number Diff line number Diff line
@@ -2394,7 +2394,7 @@ status_t AudioFlinger::createRecord(const media::CreateRecordRequest& _input, 
    CreateRecordInput input = VALUE_OR_RETURN_STATUS(CreateRecordInput::fromAidl(_input));

    CreateRecordOutput output;



    sp<RecordThread::RecordTrack> recordTrack;

    sp<IAfRecordTrack> recordTrack;

    sp<Client> client;

    status_t lStatus;

    audio_session_t sessionId = input.sessionId;
@@ -3985,7 +3985,7 @@ void AudioFlinger::updateSecondaryOutputsForTrack_l( 
            // use an index mask here to create the PatchRecord.

            inChannelMask = audio_channel_mask_out_to_in_index_mask(track->channelMask());

        }

        sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,

        sp<IAfPatchRecord> patchRecord = IAfPatchRecord::create(nullptr /* thread */,

                                                       track->sampleRate(),

                                                       inChannelMask,

                                                       track->format(),
@@ -4005,7 +4005,7 @@ void AudioFlinger::updateSecondaryOutputsForTrack_l( 
        // for now, we exclude fast tracks by removing the Fast flag.

        const audio_output_flags_t outputFlags =

                (audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);

        sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,

        sp<IAfPatchTrack> patchTrack = IAfPatchTrack::create(secondaryThread,

                                                       track->streamType(),

                                                       track->sampleRate(),

                                                       track->channelMask(),

services/audioflinger/AudioFlinger.h

+15 −8
Original line number Diff line number Diff line
@@ -119,6 +119,7 @@ 
#include "android/media/BnEffect.h"



#include "Client.h"

#include "ResamplerBufferProvider.h"



// include AudioFlinger component interfaces

#include "IAfEffect.h"
@@ -136,7 +137,6 @@ class EffectsFactoryHalInterface; 
class FastMixer;

class IAudioManager;

class PassthruBufferProvider;

class RecordBufferConverter;

class ServerProxy;



// ----------------------------------------------------------------------------
@@ -571,8 +571,6 @@ public: 
    class AsyncCallbackThread;

    class BitPerfectThread;

private:

    class Track;

    class RecordTrack;

    class DeviceEffectManager;

    // TODO(b/288339104) these should be separate files

public:
@@ -581,8 +579,9 @@ public: 
private:

    struct AudioStreamIn;

    struct TeePatch;

public:

    using TeePatches = std::vector<TeePatch>;



private:



    struct  stream_type_t {

        stream_type_t()
@@ -708,9 +707,12 @@ private: 
                                         RecordThread *srcThread,

                                         RecordThread *dstThread);



public:

    // TODO(b/288339104) cluster together

              status_t moveAuxEffectToIo(int EffectId,

                                         const sp<PlaybackThread>& dstThread,

                                         sp<PlaybackThread> *srcThread);

private:



              // return thread associated with primary hardware device, or NULL

              PlaybackThread *primaryPlaybackThread_l() const;
@@ -731,7 +733,10 @@ private: 


                void        removeClient_l(pid_t pid);

                void        removeNotificationClient(pid_t pid);

public:

    // TODO(b/288339104) cluster together

                bool isNonOffloadableGlobalEffectEnabled_l();

private:

                void onNonOffloadableGlobalEffectEnable();

                bool isSessionAcquired_l(audio_session_t audioSession);
@@ -789,8 +794,8 @@ private: 
    };



    struct TeePatch {

        sp<RecordThread::PatchRecord> patchRecord;

        sp<PlaybackThread::PatchTrack> patchTrack;

        sp<IAfPatchRecord> patchRecord;

        sp<IAfPatchTrack> patchTrack;

    };



    // for mAudioSessionRefs only
@@ -803,11 +808,13 @@ private: 
        int         mCnt;

    };



public:

    // TODO(b/288339104) access by getter,

    mutable     Mutex                               mLock;

                // protects mClients and mNotificationClients.

                // must be locked after mLock and ThreadBase::mLock if both must be locked

                // avoids acquiring AudioFlinger::mLock from inside thread loop.

public:



    // TODO(b/288339104) access by getter,

    mutable     Mutex                               mClientLock;

private:
@@ -916,9 +923,9 @@ private: 
                                      size_t rejectedKVPSize, const String8& rejectedKVPs,

                                      uid_t callingUid);



public:

    sp<IAudioManager> getOrCreateAudioManager();



public:

    // These methods read variables atomically without mLock,

    // though the variables are updated with mLock.

    bool    isLowRamDevice() const { return mIsLowRamDevice; }

services/audioflinger/IAfTrack.h

+162 −1
Original line number Diff line number Diff line
@@ -223,6 +223,9 @@ public: 
// Common interface for Playback tracks.

class IAfTrack : public virtual IAfTrackBase {

public:

    // FillingStatus is used for suppressing volume ramp at begin of playing

    enum FillingStatus { FS_INVALID, FS_FILLING, FS_FILLED, FS_ACTIVE };



    // createIAudioTrackAdapter() is a static constructor which creates an

    // IAudioTrack AIDL interface adapter from the Track object that

    // may be passed back to the client (if needed).
@@ -230,6 +233,31 @@ public: 
    // Only one AIDL IAudioTrack interface adapter should be created per Track.

    static sp<media::IAudioTrack> createIAudioTrackAdapter(const sp<IAfTrack>& track);



    static sp<IAfTrack> create( // TODO(b/288339104) void*

            void* /* AudioFlinger::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,

            bool isBitPerfect = false);



    virtual void pause() = 0;

    virtual void flush() = 0;

    virtual audio_stream_type_t streamType() const = 0;
@@ -316,11 +344,66 @@ public: 
    virtual void disable() = 0;

    virtual int& fastIndex() = 0;

    virtual bool isPlaybackRestricted() const = 0;



    // Used by thread only



    virtual bool isPausing() const = 0;

    virtual bool isPaused() const = 0;

    virtual bool isResuming() const = 0;

    virtual bool isReady() const = 0;

    virtual void setPaused() = 0;

    virtual void reset() = 0;

    virtual bool isFlushPending() const = 0;

    virtual void flushAck() = 0;

    virtual bool isResumePending() const = 0;

    virtual void resumeAck() = 0;

    // For direct or offloaded tracks ensure that the pause state is acknowledged

    // by the playback thread in case of an immediate flush.

    virtual bool isPausePending() const = 0;

    virtual void pauseAck() = 0;

    virtual void updateTrackFrameInfo(

            int64_t trackFramesReleased, int64_t sinkFramesWritten, uint32_t halSampleRate,

            const ExtendedTimestamp& timeStamp) = 0;

    virtual sp<IMemory> sharedBuffer() const = 0;



    // Dup with ExtendedAudioBufferProvider

    virtual size_t framesReady() const = 0;



    // presentationComplete checked by frames. (Mixed Tracks).

    // framesWritten is cumulative, never reset, and is shared all tracks

    // audioHalFrames is derived from output latency

    virtual bool presentationComplete(int64_t framesWritten, size_t audioHalFrames) = 0;



    // presentationComplete checked by time. (Direct Tracks).

    virtual bool presentationComplete(uint32_t latencyMs) = 0;



    virtual void resetPresentationComplete() = 0;



    virtual bool hasVolumeController() const = 0;

    virtual void setHasVolumeController(bool hasVolumeController) = 0;

    virtual const sp<AudioTrackServerProxy>& audioTrackServerProxy() const = 0;

    virtual void setCachedVolume(float volume) = 0;

    virtual void setResetDone(bool resetDone) = 0;



    virtual ExtendedAudioBufferProvider* asExtendedAudioBufferProvider() = 0;

    virtual VolumeProvider* asVolumeProvider() = 0;



    // TODO(b/288339104) split into getter/setter

    virtual FillingStatus& fillingStatus() = 0;

    virtual int8_t& retryCount() = 0;

    virtual FastTrackUnderruns& fastTrackUnderruns() = 0;

};



// playback track, used by DuplicatingThread

class IAfOutputTrack : public virtual IAfTrack {

public:

    // TODO(b/288339104) void*

    static sp<IAfOutputTrack> create(

            void* /* AudioFlinger::PlaybackThread */ playbackThread,

            void* /* AudioFlinger::DuplicatingThread */ sourceThread, uint32_t sampleRate,

            audio_format_t format, audio_channel_mask_t channelMask, size_t frameCount,

            const AttributionSourceState& attributionSource);



    virtual ssize_t write(void* data, uint32_t frames) = 0;

    virtual bool bufferQueueEmpty() const = 0;

    virtual bool isActive() const = 0;
@@ -333,6 +416,18 @@ public: 


class IAfMmapTrack : public virtual IAfTrackBase {

public:

    // TODO(b/288339104) void*

    static sp<IAfMmapTrack> create(void* /*AudioFlinger::ThreadBase */ thread,

            const audio_attributes_t& attr,

            uint32_t sampleRate,

            audio_format_t format,

            audio_channel_mask_t channelMask,

            audio_session_t sessionId,

            bool isOut,

            const android::content::AttributionSourceState& attributionSource,

            pid_t creatorPid,

            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);



    // protected by MMapThread::mLock

    virtual void setSilenced_l(bool silenced) = 0;

    // protected by MMapThread::mLock
@@ -348,6 +443,8 @@ public: 
            const sp<IAudioManager>& audioManager, mute_state_t muteState) = 0;

};



class RecordBufferConverter;



class IAfRecordTrack : public virtual IAfTrackBase {

public:

    // createIAudioRecordAdapter() is a static constructor which creates an
@@ -357,6 +454,24 @@ public: 
    // Only one AIDL IAudioRecord interface adapter should be created per RecordTrack.

    static sp<media::IAudioRecord> createIAudioRecordAdapter(const sp<IAfRecordTrack>& recordTrack);



    // TODO(b/288339104) void*

    static sp<IAfRecordTrack> create(void* /* AudioFlinger::RecordThread */ thread,

            const sp<Client>& client,

            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,

            audio_session_t sessionId,

            pid_t creatorPid,

            const AttributionSourceState& attributionSource,

            audio_input_flags_t flags,

            track_type type,

            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,

            int32_t startFrames = -1);



    // clear the buffer overflow flag

    virtual void clearOverflow() = 0;

    // set the buffer overflow flag and return previous value
@@ -386,6 +501,12 @@ public: 
    using SinkMetadatas = std::vector<record_track_metadata_v7_t>;

    using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;

    virtual void copyMetadataTo(MetadataInserter& backInserter) const = 0; // see IAfTrack



    // private to Threads

    virtual AudioBufferProvider::Buffer& sinkBuffer() = 0;

    virtual audioflinger::SynchronizedRecordState& synchronizedRecordState() = 0;

    virtual RecordBufferConverter* recordBufferConverter() const = 0;

    virtual ResamplerBufferProvider* resamplerBufferProvider() const = 0;

};



// PatchProxyBufferProvider interface is implemented by PatchTrack and PatchRecord.
@@ -401,13 +522,32 @@ public: 


class IAfPatchTrackBase : public virtual RefBase {

public:

    using Timeout = std::optional<std::chrono::nanoseconds>;



    virtual void setPeerTimeout(std::chrono::nanoseconds timeout) = 0;

    virtual void setPeerProxy(const sp<IAfPatchTrackBase>& proxy, bool holdReference) = 0;

    virtual void clearPeerProxy() = 0;

    virtual PatchProxyBufferProvider* asPatchProxyBufferProvider() = 0;

};



class IAfPatchTrack : public virtual IAfTrack, public virtual IAfPatchTrackBase {};

class IAfPatchTrack : public virtual IAfTrack, public virtual IAfPatchTrackBase {

public:

    static sp<IAfPatchTrack> create(

            void * /* PlaybackThread */ playbackThread, // TODO(b/288339104)

            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. */);

};



// Abstraction for the Audio Source for the RecordThread (HAL or PassthruPatchRecord).

struct Source {
@@ -420,6 +560,27 @@ struct Source { 


class IAfPatchRecord : public virtual IAfRecordTrack, public virtual IAfPatchTrackBase {

public:

    static sp<IAfPatchRecord> create(

            void* /* RecordThread */ recordThread, // TODO(b/288339104)

            uint32_t sampleRate,

            audio_channel_mask_t channelMask,

            audio_format_t format,

            size_t frameCount,

            void* buffer,

            size_t bufferSize,

            audio_input_flags_t flags,

            const Timeout& timeout = {},

            audio_source_t source = AUDIO_SOURCE_DEFAULT);



    static sp<IAfPatchRecord> createPassThru(

            void* /* RecordThread */ recordThread, // TODO(b/288339104)

            uint32_t sampleRate,

            audio_channel_mask_t channelMask,

            audio_format_t format,

            size_t frameCount,

            audio_input_flags_t flags,

            audio_source_t source = AUDIO_SOURCE_DEFAULT);



    virtual Source* getSource() = 0;

    virtual size_t writeFrames(const void* src, size_t frameCount, size_t frameSize) = 0;

};

services/audioflinger/MmapTracks.h

+5 −4
Original line number Diff line number Diff line
@@ -15,14 +15,14 @@ 
** limitations under the License.

*/



#ifndef INCLUDING_FROM_AUDIOFLINGER_H

    #error This header file should only be included from AudioFlinger.h

#endif

#pragma once



namespace android {



// playback track

class MmapTrack : public TrackBase, public IAfMmapTrack {

public:

                MmapTrack(ThreadBase *thread,

                MmapTrack(AudioFlinger::ThreadBase* thread,

                            const audio_attributes_t& attr,

                            uint32_t sampleRate,

                            audio_format_t format,
@@ -87,3 +87,4 @@ private: 
            GUARDED_BY(AudioFlinger::MmapPlaybackThread::mLock);

};  // end of Track



} // namespace android
 
 No newline at end of file

services/audioflinger/PatchPanel.cpp

+4 −4
Original line number Diff line number Diff line
@@ -556,7 +556,7 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel) 
        outputFlags = (audio_output_flags_t) (outputFlags & ~AUDIO_OUTPUT_FLAG_FAST);

    }



    sp<RecordThread::PatchRecord> tempRecordTrack;

    sp<IAfPatchRecord> tempRecordTrack;

    const bool usePassthruPatchRecord =

            (inputFlags & AUDIO_INPUT_FLAG_DIRECT) && (outputFlags & AUDIO_OUTPUT_FLAG_DIRECT);

    const size_t playbackFrameCount = mPlayback.thread()->frameCount();
@@ -568,7 +568,7 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel) 
        frameCount = std::max(playbackFrameCount, recordFrameCount);

        ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",

            __func__, playbackFrameCount, recordFrameCount, frameCount);

        tempRecordTrack = new RecordThread::PassthruPatchRecord(

        tempRecordTrack = IAfPatchRecord::createPassThru(

                                                 mRecord.thread().get(),

                                                 sampleRate,

                                                 inChannelMask,
@@ -587,7 +587,7 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel) 
        ALOGV("%s() playframeCount %zu recordFrameCount %zu frameCount %zu",

            __func__, playbackFrameCount, recordFrameCount, frameCount);



        tempRecordTrack = new RecordThread::PatchRecord(

        tempRecordTrack = IAfPatchRecord::create(

                                                 mRecord.thread().get(),

                                                 sampleRate,

                                                 inChannelMask,
@@ -612,7 +612,7 @@ status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel) 
    // Disable this behavior for FM Tuner source if no fast capture/mixer available.

    const bool isFmBridge = mAudioPatch.sources[0].ext.device.type == AUDIO_DEVICE_IN_FM_TUNER;

    const size_t frameCountToBeReady = isFmBridge && !usePassthruPatchRecord ? frameCount / 4 : 1;

    sp<PlaybackThread::PatchTrack> tempPatchTrack = new PlaybackThread::PatchTrack(

    sp<IAfPatchTrack> tempPatchTrack = IAfPatchTrack::create(

                                           mPlayback.thread().get(),

                                           streamType,

                                           sampleRate,