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Commit a5bea7bf authored by Andy Hung's avatar Andy Hung Committed by Automerger Merge Worker
Browse files

Merge "AudioFlinger: Add ThreadBase and PlaybackThread interfaces" into... 

Merge "AudioFlinger: Add ThreadBase and PlaybackThread interfaces" into udc-dev-plus-aosp am: 781fdf36

Original change: https://googleplex-android-review.googlesource.com/c/platform/frameworks/av/+/24473111



Change-Id: Idd6bb146200f524a58c54c796b7b8a0808e02e73
Signed-off-by: default avatarAutomerger Merge Worker <android-build-automerger-merge-worker@system.gserviceaccount.com>
parents f3bf0e9a 781fdf36

services/audioflinger/AudioFlinger.h

+1 −10
Original line number Diff line number Diff line
@@ -124,6 +124,7 @@ 


// include AudioFlinger component interfaces

#include "IAfEffect.h"

#include "IAfThread.h"

#include "IAfTrack.h"



namespace android {
@@ -562,16 +563,6 @@ private: 
    void requestLogMerge();



    // TODO(b/288339104) replace these forward declaration classes with interfaces.

public:

    class RecordThread;

    class PlaybackThread;

    class MixerThread;

    class DirectOutputThread;

    class OffloadThread;

    class DuplicatingThread;

    class AsyncCallbackThread;

    class BitPerfectThread;

private:

    class DeviceEffectManager;

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

public:

services/audioflinger/IAfThread.h

0 → 100644
+334 −0
Original line number Diff line number Diff line 
/*

 * Copyright (C) 2023 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.

 */



#pragma once



#include "IAfTrack.h"



namespace android {



class IAfThreadBase : public virtual RefBase {

public:

    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_PLAYBACK,  // Thread class for MMAP playback stream

        MMAP_CAPTURE,   // Thread class for MMAP capture stream

        SPATIALIZER,    //

        BIT_PERFECT,    // Thread class for BitPerfectThread

        // When adding a value, also update IAfThreadBase::threadTypeToString()

    };



    static const char* threadTypeToString(type_t type);

    virtual status_t readyToRun() = 0;

    virtual void clearPowerManager() = 0;

    virtual status_t initCheck() const = 0;

    virtual type_t type() const = 0;

    virtual bool isDuplicating() const = 0;

    virtual audio_io_handle_t id() const = 0;

    virtual uint32_t sampleRate() const = 0;

    virtual audio_channel_mask_t channelMask() const = 0;

    virtual audio_channel_mask_t mixerChannelMask() const = 0;

    virtual audio_format_t format() const = 0;

    virtual uint32_t channelCount() const = 0;



    // 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;

    virtual audio_channel_mask_t hapticChannelMask() const = 0;

    virtual uint32_t latency_l() const = 0;

    virtual void setVolumeForOutput_l(float left, float right) const = 0;



    // Return's the HAL's frame count i.e. fast mixer buffer size.

    virtual size_t frameCountHAL() const = 0;

    virtual size_t frameSize() const = 0;

    // Should be "virtual status_t requestExitAndWait()" and override same

    // method in Thread, but Thread::requestExitAndWait() is not yet virtual.

    virtual void exit() = 0;

    virtual bool checkForNewParameter_l(const String8& keyValuePair, status_t& status) = 0;

    virtual status_t setParameters(const String8& keyValuePairs) = 0;

    virtual String8 getParameters(const String8& keys) = 0;

    virtual void ioConfigChanged(

            audio_io_config_event_t 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<ConfigEvent>& event);

    virtual void sendIoConfigEvent(

            audio_io_config_event_t event, pid_t pid = 0,

            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;

    virtual void sendIoConfigEvent_l(

            audio_io_config_event_t event, pid_t pid = 0,

            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;

    virtual void sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;

    virtual void sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp) = 0;

    virtual status_t sendSetParameterConfigEvent_l(const String8& keyValuePair) = 0;

    virtual status_t sendCreateAudioPatchConfigEvent(

            const struct audio_patch* patch, audio_patch_handle_t* handle) = 0;

    virtual status_t sendReleaseAudioPatchConfigEvent(audio_patch_handle_t handle) = 0;

    virtual status_t sendUpdateOutDeviceConfigEvent(

            const DeviceDescriptorBaseVector& outDevices) = 0;

    virtual void sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs) = 0;

    virtual void sendCheckOutputStageEffectsEvent() = 0;

    virtual void sendCheckOutputStageEffectsEvent_l() = 0;

    virtual void sendHalLatencyModesChangedEvent_l() = 0;



    virtual void processConfigEvents_l() = 0;

    virtual void setCheckOutputStageEffects() = 0;

    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 updateOutDevices(const DeviceDescriptorBaseVector& outDevices) = 0;

    virtual void toAudioPortConfig(struct audio_port_config* config) = 0;

    virtual void resizeInputBuffer_l(int32_t maxSharedAudioHistoryMs) = 0;



    // see note at declaration of mStandby, mOutDevice and mInDevice

    virtual bool inStandby() const = 0;

    virtual const DeviceTypeSet outDeviceTypes() const = 0;

    virtual audio_devices_t inDeviceType() const = 0;

    virtual DeviceTypeSet getDeviceTypes() const = 0;

    virtual const AudioDeviceTypeAddrVector& outDeviceTypeAddrs() const = 0;

    virtual const AudioDeviceTypeAddr& inDeviceTypeAddr() const = 0;

    virtual bool isOutput() const = 0;

    virtual bool isOffloadOrMmap() const = 0;

    virtual sp<StreamHalInterface> stream() const = 0;

    virtual sp<IAfEffectHandle> createEffect_l(

            const sp<Client>& client,

            const sp<media::IEffectClient>& effectClient,

            int32_t priority,

            audio_session_t sessionId,

            effect_descriptor_t* desc,

            int* enabled,

            status_t* status /*non-NULL*/,

            bool pinned,

            bool probe,

            bool notifyFramesProcessed) = 0;



    // 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

        SPATIALIZED_SESSION = 0x8,  // the audio session corresponds to at least one

                                    // spatialized track

        BIT_PERFECT_SESSION = 0x10  // the audio session corresponds to at least one

                                    // bit-perfect track

    };



    // get effect chain corresponding to session Id.

    virtual sp<IAfEffectChain> getEffectChain(audio_session_t sessionId) const = 0;

    // same as getEffectChain() but must be called with ThreadBase mutex locked

    virtual sp<IAfEffectChain> getEffectChain_l(audio_session_t sessionId) const = 0;

    virtual std::vector<int> getEffectIds_l(audio_session_t sessionId) const = 0;

    // add an effect chain to the chain list (mEffectChains)

    virtual status_t addEffectChain_l(const sp<IAfEffectChain>& chain) = 0;

    // remove an effect chain from the chain list (mEffectChains)

    virtual size_t removeEffectChain_l(const sp<IAfEffectChain>& 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.

    virtual void lockEffectChains_l(Vector<sp<IAfEffectChain>>& effectChains) = 0;

    // unlock effect chains after process

    virtual void unlockEffectChains(const Vector<sp<IAfEffectChain>>& effectChains) = 0;

    // get a copy of mEffectChains vector

    virtual Vector<sp<IAfEffectChain>> getEffectChains_l() const = 0;

    // set audio mode to all effect chains

    virtual void setMode(audio_mode_t mode) = 0;

    // get effect module with corresponding ID on specified audio session

    virtual sp<IAfEffectModule> getEffect(audio_session_t sessionId, int effectId) const = 0;

    virtual sp<IAfEffectModule> getEffect_l(audio_session_t sessionId, int effectId) const = 0;

    // 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

    virtual status_t addEffect_l(const sp<IAfEffectModule>& effect) = 0;

    // remove and effect module. Also removes the effect chain is this was the last

    // effect

    virtual void removeEffect_l(const sp<IAfEffectModule>& effect, bool release = false) = 0;

    // disconnect an effect handle from module and destroy module if last handle

    virtual void disconnectEffectHandle(IAfEffectHandle* handle, bool unpinIfLast) = 0;

    // detach all tracks connected to an auxiliary effect

    virtual void detachAuxEffect_l(int effectId) = 0;

    // 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

    // - SPATIALIZED_SESSION if spatialized tracks on this audio session exist

    virtual uint32_t hasAudioSession_l(audio_session_t sessionId) const = 0;

    virtual uint32_t hasAudioSession(audio_session_t sessionId) const = 0;



    // the value returned by default implementation is not important as the

    // strategy is only meaningful for PlaybackThread which implements this method

    virtual product_strategy_t getStrategyForSession_l(audio_session_t sessionId) const = 0;



    // check if some effects must be suspended/restored when an effect is enabled

    // or disabled

    virtual void checkSuspendOnEffectEnabled(

            bool enabled, audio_session_t sessionId, bool threadLocked) = 0;



    virtual status_t setSyncEvent(const sp<audioflinger::SyncEvent>& event) = 0;

    virtual bool isValidSyncEvent(const sp<audioflinger::SyncEvent>& 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<MemoryDealer> readOnlyHeap() const = 0;



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



    virtual void systemReady() = 0;



    // checkEffectCompatibility_l() must be called with ThreadBase::mLock held

    virtual status_t checkEffectCompatibility_l(

            const effect_descriptor_t* desc, audio_session_t sessionId) = 0;



    virtual void broadcast_l() = 0;



    virtual bool isTimestampCorrectionEnabled() const = 0;



    virtual bool isMsdDevice() const = 0;



    virtual void dump(int fd, const Vector<String16>& args) = 0;



    // deliver stats to mediametrics.

    virtual void sendStatistics(bool force) = 0;



    virtual Mutex& mutex() const = 0;



    virtual void onEffectEnable(const sp<IAfEffectModule>& effect) = 0;

    virtual void onEffectDisable() = 0;



    // invalidateTracksForAudioSession_l must be called with holding mLock.

    virtual void invalidateTracksForAudioSession_l(audio_session_t sessionId) const = 0;

    // Invalidate all the tracks with the given audio session.

    virtual void invalidateTracksForAudioSession(audio_session_t sessionId) const = 0;



    virtual bool isStreamInitialized() const = 0;

    virtual void startMelComputation_l(const sp<audio_utils::MelProcessor>& processor) = 0;

    virtual void stopMelComputation_l() = 0;



    virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) const = 0;

};



class IAfPlaybackThread : public virtual IAfThreadBase {

public:

    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

    };



    // return estimated latency in milliseconds, as reported by HAL

    virtual uint32_t latency() const = 0;  // should be in IAfThreadBase?



    virtual sp<IAfTrack> createTrack_l(

            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* pFrameCount,

            size_t* pNotificationFrameCount,

            uint32_t notificationsPerBuffer,

            float speed,

            const sp<IMemory>& sharedBuffer,

            audio_session_t sessionId,

            audio_output_flags_t* flags,

            pid_t creatorPid,

            const AttributionSourceState& attributionSource,

            pid_t tid,

            status_t* status /*non-NULL*/,

            audio_port_handle_t portId,

            const sp<media::IAudioTrackCallback>& callback,

            bool isSpatialized,

            bool isBitPerfect) = 0;



    virtual AudioStreamOut* getOutput() const = 0;

    virtual AudioStreamOut* clearOutput() = 0;

    virtual sp<StreamHalInterface> stream() const = 0;



    // a very large number of suspend() will eventually wraparound, but unlikely

    virtual void suspend() = 0;

    virtual void restore() = 0;

    virtual bool isSuspended() const = 0;

    virtual status_t getRenderPosition(uint32_t* halFrames, uint32_t* dspFrames) const = 0;

    // Consider also removing and passing an explicit mMainBuffer initialization

    // parameter to AF::IAfTrack::Track().

    virtual float* sinkBuffer() const = 0;



    virtual status_t attachAuxEffect(const sp<IAfTrack>& track, int EffectId) = 0;

    virtual status_t attachAuxEffect_l(const sp<IAfTrack>& track, int EffectId) = 0;



    // called with AudioFlinger lock held

    virtual bool invalidateTracks_l(audio_stream_type_t streamType) = 0;

    virtual bool invalidateTracks_l(std::set<audio_port_handle_t>& portIds) = 0;

    virtual void invalidateTracks(audio_stream_type_t streamType) = 0;

    // Invalidate tracks by a set of port ids. The port id will be removed from

    // the given set if the corresponding track is found and invalidated.

    virtual void invalidateTracks(std::set<audio_port_handle_t>& portIds) = 0;



    virtual status_t getTimestamp_l(AudioTimestamp& timestamp) = 0;

    virtual void addPatchTrack(const sp<IAfPatchTrack>& track) = 0;

    virtual void deletePatchTrack(const sp<IAfPatchTrack>& track) = 0;



    // Return the asynchronous signal wait time.

    virtual int64_t computeWaitTimeNs_l() const = 0;

    // returns true if the track is allowed to be added to the thread.

    virtual bool isTrackAllowed_l(

            audio_channel_mask_t channelMask, audio_format_t format, audio_session_t sessionId,

            uid_t uid) const = 0;



    virtual bool supportsHapticPlayback() const = 0;



    virtual void setDownStreamPatch(const struct audio_patch* patch) = 0;



    virtual IAfTrack* getTrackById_l(audio_port_handle_t trackId) = 0;



    virtual bool hasMixer() const = 0;



    virtual status_t setRequestedLatencyMode(audio_latency_mode_t mode) = 0;



    virtual status_t getSupportedLatencyModes(std::vector<audio_latency_mode_t>* modes) = 0;



    virtual status_t setBluetoothVariableLatencyEnabled(bool enabled) = 0;



    virtual void setStandby() = 0;

    virtual void setStandby_l() = 0;

    virtual bool waitForHalStart() = 0;



    virtual bool hasFastMixer() const = 0;

    virtual FastTrackUnderruns getFastTrackUnderruns(size_t fastIndex) const = 0;

    virtual const std::atomic<int64_t>& framesWritten() const = 0;

};



}  // namespace android

services/audioflinger/Threads.cpp

+11 −10
Original line number Diff line number Diff line
@@ -1692,20 +1692,20 @@ void AudioFlinger::ThreadBase::onEffectDisable() { 
}



sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId,

        int effectId)

        int effectId) const

{

    Mutex::Autolock _l(mLock);

    return getEffect_l(sessionId, effectId);

}



sp<IAfEffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId,

        int effectId)

        int effectId) const

{

    sp<IAfEffectChain> chain = getEffectChain_l(sessionId);

    return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;

}



std::vector<int> AudioFlinger::ThreadBase::getEffectIds_l(audio_session_t sessionId)

std::vector<int> AudioFlinger::ThreadBase::getEffectIds_l(audio_session_t sessionId) const

{

    sp<IAfEffectChain> chain = getEffectChain_l(sessionId);

    return chain != nullptr ? chain->getEffectIds() : std::vector<int>{};
@@ -1796,7 +1796,7 @@ NO_THREAD_SAFETY_ANALYSIS // calls EffectChain::unlock() 
    }

}



sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId)

sp<IAfEffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) const

{

    Mutex::Autolock _l(mLock);

    return getEffectChain_l(sessionId);
@@ -3261,7 +3261,8 @@ void AudioFlinger::PlaybackThread::sendMetadataToBackend_l( 
    mOutput->stream->updateSourceMetadata(metadata);

};



status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)

status_t AudioFlinger::PlaybackThread::getRenderPosition(

        uint32_t* halFrames, uint32_t* dspFrames) const

{

    if (halFrames == NULL || dspFrames == NULL) {

        return BAD_VALUE;
@@ -3288,7 +3289,8 @@ status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, ui 
    }

}



product_strategy_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId)

product_strategy_t AudioFlinger::PlaybackThread::getStrategyForSession_l(

        audio_session_t sessionId) const

{

    // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that

    // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
@@ -7614,7 +7616,7 @@ bool AudioFlinger::DuplicatingThread::outputsReady() 
        }

        PlaybackThread *playbackThread = (PlaybackThread *)thread.get();

        // see note at standby() declaration

        if (playbackThread->standby() && !playbackThread->isSuspended()) {

        if (playbackThread->inStandby() && !playbackThread->isSuspended()) {

            ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),

                    thread.get());

            return false;
@@ -10825,7 +10827,7 @@ void AudioFlinger::MmapPlaybackThread::toAudioPortConfig(struct audio_port_confi 
}



status_t AudioFlinger::MmapPlaybackThread::getExternalPosition(uint64_t *position,

                                                               int64_t *timeNanos)

        int64_t* timeNanos) const

{

    if (mOutput == nullptr) {

        return NO_INIT;
@@ -10910,7 +10912,6 @@ AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput() 
    return input;

}





void AudioFlinger::MmapCaptureThread::processVolume_l()

{

    bool changed = false;
@@ -10983,7 +10984,7 @@ void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config 
}



status_t AudioFlinger::MmapCaptureThread::getExternalPosition(

        uint64_t *position, int64_t *timeNanos)

        uint64_t* position, int64_t* timeNanos) const

{

    if (mInput == nullptr) {

        return NO_INIT;

services/audioflinger/Threads.h

+304 −382

File changed.

Preview size limit exceeded, changes collapsed.

services/audioflinger/Tracks.cpp

+2 −2
Original line number Diff line number Diff line
@@ -2206,7 +2206,7 @@ ssize_t OutputTrack::write(void* data, uint32_t frames) 
{

    if (!mActive && frames != 0) {

        sp<AudioFlinger::ThreadBase> thread = mThread.promote();

        if (thread != nullptr && thread->standby()) {

        if (thread != nullptr && thread->inStandby()) {

            // preload one silent buffer to trigger mixer on start()

            ClientProxy::Buffer buf { .mFrameCount = mClientProxy->getStartThresholdInFrames() };

            status_t status = mClientProxy->obtainBuffer(&buf);
@@ -2314,7 +2314,7 @@ ssize_t OutputTrack::write(void* data, uint32_t frames) 
    // If we could not write all frames, allocate a buffer and queue it for next time.

    if (inBuffer.frameCount) {

        sp<AudioFlinger::ThreadBase> thread = mThread.promote();

        if (thread != 0 && !thread->standby()) {

        if (thread != nullptr && !thread->inStandby()) {

            queueBuffer(inBuffer);

        }

    }