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Android8.0 Audio系统之AudioFlinger_android audio flinger

作者：我家小花儿 | 2024-03-26 18:42:40

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android audio flinger

继上一篇AudioTrack的分析，本篇我们来看AudioFlinger，AF主要承担音频混合输出，是Audio系统的核心，从AudioTrack来的数据最终都会在这里处理，并被写入到Audio的HAL。

1. AudioFlinger 创建

frameworks\av\services\audioflinger\AudioFlinger.cpp

AudioFlinger::AudioFlinger()
    : BnAudioFlinger(),
      mMediaLogNotifier(new AudioFlinger::MediaLogNotifier()),
      mPrimaryHardwareDev(NULL),
      mAudioHwDevs(NULL),
      mHardwareStatus(AUDIO_HW_IDLE),
      mMasterVolume(1.0f),
      mMasterMute(false),
      // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX),
      mMode(AUDIO_MODE_INVALID),
      mBtNrecIsOff(false),
      mIsLowRamDevice(true),
      mIsDeviceTypeKnown(false),
      mGlobalEffectEnableTime(0),
      mSystemReady(false)
{
    ......
    mDevicesFactoryHal = DevicesFactoryHalInterface::create();
    mEffectsFactoryHal = EffectsFactoryHalInterface::create();
    ......
}

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frameworks\av\media\libaudiohal\include\DevicesFactoryHalInterface.h
其中 DevicesFactoryHalHidl.cpp , DevicesFactoryHalHybrid.cpp , DevicesFactoryHalLocal.cpp分别都继承自DevicesFactoryHalInterface

class DevicesFactoryHalInterface : public RefBase
{
  public:
    // Opens a device with the specified name. To close the device, it is
    // necessary to release references to the returned object.
    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device) = 0;

    static sp<DevicesFactoryHalInterface> create();

  protected:
    // Subclasses can not be constructed directly by clients.
    DevicesFactoryHalInterface() {}

    virtual ~DevicesFactoryHalInterface() {}
};

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在DevicesFactoryHalHybrid.cpp文件中找到了create()函数的实现

sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
    return new DevicesFactoryHalHybrid();
}

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DevicesFactoryHalHybrid.h文件中定义了mLocalFactory，mHidlFactory

class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
{
  public:
    // Opens a device with the specified name. To close the device, it is
    // necessary to release references to the returned object.
    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);

  private:
    friend class DevicesFactoryHalInterface;

    // Can not be constructed directly by clients.
    DevicesFactoryHalHybrid();

    virtual ~DevicesFactoryHalHybrid();

    sp<DevicesFactoryHalInterface> mLocalFactory;
    sp<DevicesFactoryHalInterface> mHidlFactory;
};

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DevicesFactoryHalHybrid.cpp中分别创建了不同的实现对象，通过 USE_LEGACY_LOCAL_AUDIO_HAL

DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
        : mLocalFactory(new DevicesFactoryHalLocal()),
          mHidlFactory(
#ifdef USE_LEGACY_LOCAL_AUDIO_HAL
                  nullptr
#else
                  new DevicesFactoryHalHidl()
#endif
                       ) {
}

status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0) {
        return mHidlFactory->openDevice(name, device);
    }
    return mLocalFactory->openDevice(name, device);
}

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如果是本地模式，DevicesFactoryHalLocal.cpp中使用8.0以前的HAL加载方式

static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
    const hw_module_t *mod;
    int rc;

    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
    if (rc) {
        ALOGE("%s couldn't load audio hw module %s.%s (%s)", __func__,
                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
        goto out;
    }
    rc = audio_hw_device_open(mod, dev);
    if (rc) {
        ALOGE("%s couldn't open audio hw device in %s.%s (%s)", __func__,
                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
        goto out;
    }
    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
        rc = BAD_VALUE;
        audio_hw_device_close(*dev);
        goto out;
    }
    return OK;

out:
    *dev = NULL;
    return rc;
}

status_t DevicesFactoryHalLocal::openDevice(const char *name, sp<DeviceHalInterface> *device) {
    audio_hw_device_t *dev;
    status_t rc = load_audio_interface(name, &dev);
    if (rc == OK) {
        *device = new DeviceHalLocal(dev);
    }
    return rc;
}

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如果是HIDL的Treble模式，将采用8.0的新架构

DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
    mDevicesFactory = IDevicesFactory::getService(); //后续会讲到
    if (mDevicesFactory != 0) {
        // It is assumed that DevicesFactory is owned by AudioFlinger
        // and thus have the same lifespan.
        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
    } else {
        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
        exit(1);
    }
}

DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
}

// static
status_t DevicesFactoryHalHidl::nameFromHal(const char *name, IDevicesFactory::Device *device) {
    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
        *device = IDevicesFactory::Device::PRIMARY;
        return OK;
    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
        *device = IDevicesFactory::Device::A2DP;
        return OK;
    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
        *device = IDevicesFactory::Device::USB;
        return OK;
    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
        *device = IDevicesFactory::Device::R_SUBMIX;
        return OK;
    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) == 0) {
        *device = IDevicesFactory::Device::STUB;
        return OK;
    }
    ALOGE("Invalid device name %s", name);
    return BAD_VALUE;
}

status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
    if (mDevicesFactory == 0) return NO_INIT;
    IDevicesFactory::Device hidlDevice;
    status_t status = nameFromHal(name, &hidlDevice);
    if (status != OK) return status;
    Result retval = Result::NOT_INITIALIZED;
    Return<void> ret = mDevicesFactory->openDevice(
            hidlDevice,
            [&](Result r, const sp<IDevice>& result) {
                retval = r;
                if (retval == Result::OK) {
                    *device = new DeviceHalHidl(result); //后续讲解
                }
            });
    if (ret.isOk()) {
        if (retval == Result::OK) return OK;
        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
        else return NO_INIT;
    }
    return FAILED_TRANSACTION;
}

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2. createTrack()

我们暂且跳过Audio的硬件抽象层，回头看看AudioTrack调用AF创建Track的过程

sp<IAudioTrack> AudioFlinger::createTrack(
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t *frameCount,
        audio_output_flags_t *flags,
        const sp<IMemory>& sharedBuffer,
        audio_io_handle_t output,
        pid_t pid,
        pid_t tid,
        audio_session_t *sessionId,
        int clientUid,
        status_t *status,
        audio_port_handle_t portId)
{
    sp<PlaybackThread::Track> track;
    sp<TrackHandle> trackHandle;
    sp<Client> client;
    status_t lStatus;
    audio_session_t lSessionId;

    {
        Mutex::Autolock _l(mLock);
        PlaybackThread *thread = checkPlaybackThread_l(output); //创建回放线程
        if (thread == NULL) {
            ALOGE("no playback thread found for output handle %d", output);
            lStatus = BAD_VALUE;
            goto Exit;
        }

        client = registerPid(pid); //注册到Client

        PlaybackThread *effectThread = NULL;
        if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) {
        ......
        track = thread->createTrack_l(client, streamType, sampleRate, format,
                channelMask, frameCount, sharedBuffer, lSessionId, flags, tid,
                clientUid, &lStatus, portId);
        setAudioHwSyncForSession_l(thread, lSessionId);
    }

    // return handle to client
    trackHandle = new TrackHandle(track); //返回Track的代理

Exit:
    *status = lStatus;
    return trackHandle;
}

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为Track取一个线程，并没有创建，奇怪

// checkPlaybackThread_l() 取出线程
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
    return mPlaybackThreads.valueFor(output).get(); //那么线程在哪里创建的
}

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frameworks\av\services\audioflinger\Threads.cpp

sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
        const sp<AudioFlinger::Client>& client,
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t *pFrameCount,
        const sp<IMemory>& sharedBuffer,
        audio_session_t sessionId,
        audio_output_flags_t *flags,
        pid_t tid,
        uid_t uid,
        status_t *status,
        audio_port_handle_t portId)
{
       //创建Track
       track = new Track(this, client, streamType, sampleRate, format,
                          channelMask, frameCount, NULL, sharedBuffer,
                          sessionId, uid, *flags, TrackBase::TYPE_DEFAULT, portId);
                          
    return track;
}

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那么上面的线程是如何创建出来的，我们得到APS里去看看

void AudioPolicyService::onFirstRef()
{
    {
        Mutex::Autolock _l(mLock);

        // start tone playback thread
        mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
        // start audio commands thread
        mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
        // start output activity command thread
        mOutputCommandThread = new AudioCommandThread(String8("ApmOutput"), this);

        mAudioPolicyClient = new AudioPolicyClient(this);
        mAudioPolicyManager = createAudioPolicyManager(mAudioPolicyClient); //创建代理管家
    }
    // load audio processing modules
    sp<AudioPolicyEffects>audioPolicyEffects = new AudioPolicyEffects();
    {
        Mutex::Autolock _l(mLock);
        mAudioPolicyEffects = audioPolicyEffects;
    }
}

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frameworks\av\services\audiopolicy\manager\AudioPolicyFactory.cpp

extern "C" AudioPolicyInterface* createAudioPolicyManager(
        AudioPolicyClientInterface *clientInterface)
{
    return new AudioPolicyManager(clientInterface);
}

extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
{
    delete interface;
}

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frameworks\av\services\audiopolicy\managerdefault\AudioPolicyManager.cpp

udioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
   ......
{
     ......
    //调用APS的openOutput
    status_t status = mpClientInterface->openOutput(......);
}

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status_t AudioPolicyService::AudioPolicyClient::openOutput(......)
{
    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
    if (af == 0) {
        ALOGW("%s: could not get AudioFlinger", __func__);
        return PERMISSION_DENIED;
    }
    return af->openOutput(module, output, config, devices, address, latencyMs, flags);
}


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调用AF的openOutput

status_t AudioFlinger::openOutput(......)
{
    ......
    sp<ThreadBase> thread = openOutput_l(module, output, config, *devices, address, flags);
    ......
    return NO_INIT;
}

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饶了一圈回来在此创建线程，并放入到mPlaybackThreads中

sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(......)
{
     sp<PlaybackThread> thread;
     if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
            thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady);
        } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
                    || !isValidPcmSinkFormat(config->format)
                    || !isValidPcmSinkChannelMask(config->channel_mask)) {
            thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady);
        } else {
            thread = new MixerThread(this, outputStream, *output, devices, mSystemReady);
        }
        mPlaybackThreads.add(*output, thread);
        return thread;
}


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3. start()

frameworks\av\services\audioflinger\Tracks.cpp
由此可见APS控制着整个音频系统，而AF管理的是音频输入混合输出，AudioTrack创建成功以后，开始调用start()函数，start()函数将由TrackHandle代理并调用Track的start()函数

status_t AudioFlinger::PlaybackThread::Track::start(......)
{
    status_t status = NO_ERROR;

    sp<ThreadBase> thread = mThread.promote(); 
       if (state == PAUSED || state == PAUSING) {
            if (mResumeToStopping) {
                // happened we need to resume to STOPPING_1
                mState = TrackBase::STOPPING_1;
                ALOGV("PAUSED => STOPPING_1 (%d) on thread %p", mName, this);
            } else {
                mState = TrackBase::RESUMING;
                ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this);
            }
        } else {
            mState = TrackBase::ACTIVE; //设置Track的状态
            ALOGV("? => ACTIVE (%d) on thread %p", mName, this);
        }
        status = playbackThread->addTrack_l(this);
    return status;
}

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将创建好的Track加入到mActiveTracks中

status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
    status_t status = ALREADY_EXISTS;

    if (mActiveTracks.indexOf(track) < 0) {
        ......
        // set retry count for buffer fill
        if (track->isOffloaded()) {
            if (track->isStopping_1()) {
                track->mRetryCount = kMaxTrackStopRetriesOffload;
            } else {
                track->mRetryCount = kMaxTrackStartupRetriesOffload;
            }
            track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED;
        } else {
            track->mRetryCount = kMaxTrackStartupRetries; //重试次数
            track->mFillingUpStatus =
                    track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING;
        }

        track->mResetDone = false;
        track->mPresentationCompleteFrames = 0;
        mActiveTracks.add(track);
        sp<EffectChain> chain = getEffectChain_l(track->sessionId());
        if (chain != 0) {
            chain->incActiveTrackCnt();
        }

        char buffer[256];
        track->dump(buffer, ARRAY_SIZE(buffer), false /* active */);

        status = NO_ERROR;
    }

    onAddNewTrack_l(); //触发音频混合线程MixerThread的
    return status;
}

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PlaybackThread的threadLoop()将调用MixerThread::prepareTracks_l()函数

AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
        Vector< sp<Track> > *tracksToRemove)
{

    mixer_state mixerStatus = MIXER_IDLE;
    // find out which tracks need to be processed
    size_t count = mActiveTracks.size(); //已激活的Track数量
    ......
    for (size_t i=0 ; i<count ; i++) { //遍历查询
        const sp<Track> t = mActiveTracks[i];

        // this const just means the local variable doesn't change
        Track* const track = t.get();

        {
        //取cblk
        audio_track_cblk_t* cblk = track->cblk();

        // The first time a track is added we wait
        // for all its buffers to be filled before processing it
        int name = track->name();
       ......
      // XXX: these things DON'T need to be done each time
            //设置数据来源为Track
            mAudioMixer->setBufferProvider(name, track);
            mAudioMixer->enable(name);
            //设置音量等参数
            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf);
            mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf);
            mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf);
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::FORMAT, (void *)track->format());
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask());
            mAudioMixer->setParameter(
                name,
                AudioMixer::TRACK,
                AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask);
        
    ......
    mMixerStatusIgnoringFastTracks = mixerStatus;
    if (fastTracks > 0) {
        mixerStatus = MIXER_TRACKS_READY;
    }
    return mixerStatus;
}

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混合线程的处理函数

void AudioFlinger::MixerThread::threadLoop_mix()
{
    // mix buffers...
    mAudioMixer->process();
    if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) {
        sleepTimeShift--;
    }
    mSleepTimeUs = 0;
    mStandbyTimeNs = systemTime() + mStandbyDelayNs;
    //TODO: delay standby when effects have a tail

}

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调用混音器的处理函数

void AudioMixer::process()
{
    mState.hook(&mState); //hook指针根据Track的个数和它的音频格式使用不同的处理函数
}

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使能混合器

void AudioMixer::enable(int name)
{
    name -= TRACK0;
    ALOG_ASSERT(uint32_t(name) < MAX_NUM_TRACKS, "bad track name %d", name);
    track_t& track = mState.tracks[name];

    if (!track.enabled) {
        track.enabled = true;
        ALOGV("enable(%d)", name);
        invalidateState(1 << name); //刷新
    }
}


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刷新缓冲

void AudioMixer::invalidateState(uint32_t mask)
{
    if (mask != 0) {
        mState.needsChanged |= mask;
        mState.hook = process__validate; //赋给函数指针
    }
 }

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采样处理函数

void AudioMixer::process__validate(state_t* state)
{
    
    uint32_t enabled = 0;
    uint32_t disabled = 0;
    ......
    if (countActiveTracks > 0) {
        if (resampling) {
            .....
            state->hook = process__genericResampling; //采样
        } else {
            if (state->outputTemp) {
                delete [] state->outputTemp;
                state->outputTemp = NULL;
            }
            if (state->resampleTemp) {
                delete [] state->resampleTemp;
                state->resampleTemp = NULL;
            }
            state->hook = process_OneTrack16BitsStereoNoResampling; //采样
       }
}


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从Track中取出音频数据，写出

void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
{
    t.bufferProvider->getNextBuffer(&b); //获得可读缓冲
    numFrames -= b.frameCount;
    t.bufferProvider->releaseBuffer(&b); //释放缓冲区
}

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MixerThread获取Track的数据，进行混音后通过AudioStreamOutPut的 *output 写入音频输出设备

AudioFlinger的分析就到这里，AudioFlinger通过APS创建混音线程，混音线程将Track中的数据取出，进入环形缓冲区处理，最终输出到音频硬件设备；这个过程较为复杂，特别audio_track_cblk在此并没有详细分析；AF的分析暂且到此结束，后续连贯分析音频流程的时候会更清晰。Bye,Bye !

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