在上一篇文章《（二）Audio子系統之new AudioRecord()》中已經介紹了Audio系統如何建立AudioRecord物件以及輸入流，並建立了RecordThread執行緒，接下來，繼續分析AudioRecord方法中的startRecording的實現

　　函式原型：

     public void startRecording() throws IllegalStateException

       作用：

　　　　開始進行錄製

　　引數：

　　　　無

　　返回值：

　　　　無

      異常：

　　　　若沒有初始化完成時，丟擲IllegalStateException

接下來進入系統分析具體實現

frameworks/base/media/java/android/media/AudioRecord.java

    public void startRecording()
    throws IllegalStateException {
        if (mState != STATE_INITIALIZED) {
            throw new IllegalStateException("startRecording() called on an "
                    + "uninitialized AudioRecord.");
        }

        // start recording
        synchronized(mRecordingStateLock) {
            if (native_start(MediaSyncEvent.SYNC_EVENT_NONE, 0) == SUCCESS) {
                handleFullVolumeRec(true);
                mRecordingState = RECORDSTATE_RECORDING;
            }
        }
    }

首先判斷是否已經初始化完畢了，在前一篇文章中，mState已經是STATE_INITIALIZED狀態了。所以繼續分析native_start函式

frameworks/base/core/jni/android_media_AudioRecord.cpp

static jint
android_media_AudioRecord_start(JNIEnv *env, jobject thiz, jint event, jint triggerSession)
{
    sp<AudioRecord> lpRecorder = getAudioRecord(env, thiz);
    if (lpRecorder == NULL ) {
        jniThrowException(env, "java/lang/IllegalStateException", NULL);
        return (jint) AUDIO_JAVA_ERROR;
    }

    return nativeToJavaStatus(
            lpRecorder->start((AudioSystem::sync_event_t)event, triggerSession));
}

繼續往下：lpRecorder->start

frameworks\av\media\libmedia\AudioRecord.cpp

status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
{
    AutoMutex lock(mLock);
    if (mActive) {
        return NO_ERROR;
    }

    // reset current position as seen by client to 0
    mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
    // force refresh of remaining frames by processAudioBuffer() as last
    // read before stop could be partial.
    mRefreshRemaining = true;

    mNewPosition = mProxy->getPosition() + mUpdatePeriod;
	
    int32_t flags = android_atomic_acquire_load(&mCblk->mFlags);

    status_t status = NO_ERROR;
    if (!(flags & CBLK_INVALID)) {
        ALOGV("mAudioRecord->start()");
        status = mAudioRecord->start(event, triggerSession);
        if (status == DEAD_OBJECT) {
            flags |= CBLK_INVALID;
        }
    }
    if (flags & CBLK_INVALID) {
        status = restoreRecord_l("start");	
    }

    if (status != NO_ERROR) {
        ALOGE("start() status %d", status);
    } else {
        mActive = true;
        sp<AudioRecordThread> t = mAudioRecordThread;
        if (t != 0) {
            t->resume();
        } else {
            mPreviousPriority = getpriority(PRIO_PROCESS, 0);
            get_sched_policy(0, &mPreviousSchedulingGroup);
            androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
        }
    }

    return status;
}

在這個函式中主要的工作如下：

    1.重置當前錄音Buffer中的錄音資料寫入的起始位置，錄音Buffer的組成在第一篇文章中已經介紹了；

    2.標記mRefreshRemaining為true，從註釋中可以看到，他應該是用來強制重新整理剩餘的frames，後面應該會突出這個變數的作用，先不急；

    3.從mCblk->mFlags的地方獲取flags，這裡是0x0；

    4.第一次來，肯定走mAudioRecord->start()；

    5.如果start失敗了，會重新呼叫restoreRecord_l函式，再次建立輸入流通道，這個函式在前一篇文章已經分析過了；

    6.呼叫AudioRecordThread執行緒的resume函式；

這裡我們主要分析第4、6步；

首先分析下AudioRecord.cpp::start()的第4步：mAudioRecord->start()

mAudioRecord是sp<IAudioRecord>型別的，也就是說他是Binder中的Bp端，我們需要找到BnAudioRecord，可以在AudioFlinger.h中找到Bn端的定義

frameworks\av\services\audioflinger\AudioFlinger.h

    // server side of the client's IAudioRecord
    class RecordHandle : public android::BnAudioRecord {
    public:
        RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
        virtual             ~RecordHandle();
        virtual status_t    start(int /*AudioSystem::sync_event_t*/ event, int triggerSession);
        virtual void        stop();
        virtual status_t onTransact(
            uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
    private:
        const sp<RecordThread::RecordTrack> mRecordTrack;

        // for use from destructor
        void                stop_nonvirtual();
    };

所以我們繼續找RecordHandle類是在哪裡實現的，同時，這裡可以看到除了start方法以外還有stop方法。

frameworks\av\services\audioflinger\Tracks.cpp

status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
        int triggerSession) {

    return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
}

在AudioFlinger.h檔案中可以看到const sp<RecordThread::RecordTrack> mRecordTrack;他還是在Tracks.cpp中實現的，繼續往下走

status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event,
                                                        int triggerSession)
{

    sp<ThreadBase> thread = mThread.promote();
    if (thread != 0) {
        RecordThread *recordThread = (RecordThread *)thread.get();
        return recordThread->start(this, event, triggerSession);
    } else {
        return BAD_VALUE;
    }
}

這裡的Thread是在AudioRecord.cpp::openRecord_l()中呼叫createRecordTrack_l的Thread物件，再深入一點，在thread->createRecordTrack_l方法中呼叫了new RecordTrack(this,...)，而RecordTrack是繼承TrackBase的，在TrackBase父類的建構函式中TrackBase(ThreadBase *thread,...): RefBase(), mThread(thread),...{}，這個父類的實現也是在Tracks.cpp。所以這裡的mThread就是RecordThread

所以這裡繼續呼叫RecordThread的start方法

frameworks\av\services\audioflinger\Threads.cpp

status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack,
                                           AudioSystem::sync_event_t event,
                                           int triggerSession)
{
    sp<ThreadBase> strongMe = this;
    status_t status = NO_ERROR;

    if (event == AudioSystem::SYNC_EVENT_NONE) {
        recordTrack->clearSyncStartEvent();
    } else if (event != AudioSystem::SYNC_EVENT_SAME) {
        recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event,
                                       triggerSession,
                                       recordTrack->sessionId(),
                                       syncStartEventCallback,
                                       recordTrack);
        // Sync event can be cancelled by the trigger session if the track is not in a
        // compatible state in which case we start record immediately
        if (recordTrack->mSyncStartEvent->isCancelled()) {
            recordTrack->clearSyncStartEvent();
        } else {
            // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs
            recordTrack->mFramesToDrop = -
                    ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000);
        }
    }

    {
        // This section is a rendezvous between binder thread executing start() and RecordThread
        AutoMutex lock(mLock);
        if (mActiveTracks.indexOf(recordTrack) >= 0) {
            if (recordTrack->mState == TrackBase::PAUSING) {
                ALOGV("active record track PAUSING -> ACTIVE");
                recordTrack->mState = TrackBase::ACTIVE;
            } else {
                ALOGV("active record track state %d", recordTrack->mState);
            }
            return status;
        }

        // TODO consider other ways of handling this, such as changing the state to :STARTING and
        //      adding the track to mActiveTracks after returning from AudioSystem::startInput(),
        //      or using a separate command thread
        recordTrack->mState = TrackBase::STARTING_1;
        mActiveTracks.add(recordTrack);
        mActiveTracksGen++;
        status_t status = NO_ERROR;
        if (recordTrack->isExternalTrack()) {
            mLock.unlock();
            status = AudioSystem::startInput(mId, (audio_session_t)recordTrack->sessionId());
            mLock.lock();
            // FIXME should verify that recordTrack is still in mActiveTracks
            if (status != NO_ERROR) {//0
                mActiveTracks.remove(recordTrack);
                mActiveTracksGen++;
                recordTrack->clearSyncStartEvent();
                ALOGV("RecordThread::start error %d", status);
                return status;
            }
        }
        // Catch up with current buffer indices if thread is already running.
        // This is what makes a new client discard all buffered data.  If the track's mRsmpInFront
        // was initialized to some value closer to the thread's mRsmpInFront, then the track could
        // see previously buffered data before it called start(), but with greater risk of overrun.

        recordTrack->mRsmpInFront = mRsmpInRear;
        recordTrack->mRsmpInUnrel = 0;
        // FIXME why reset?
        if (recordTrack->mResampler != NULL) {
            recordTrack->mResampler->reset();
        }
        recordTrack->mState = TrackBase::STARTING_2;
        // signal thread to start
        mWaitWorkCV.broadcast();
        if (mActiveTracks.indexOf(recordTrack) < 0) {
            ALOGV("Record failed to start");
            status = BAD_VALUE;
            goto startError;
        }
        return status;
    }

startError:
    if (recordTrack->isExternalTrack()) {
        AudioSystem::stopInput(mId, (audio_session_t)recordTrack->sessionId());
    }
    recordTrack->clearSyncStartEvent();
    // FIXME I wonder why we do not reset the state here?
    return status;
}

在這個函式中主要的工作如下：

    1.判斷傳過來的event的值，從AudioRecord.java可以看到他一直是SYNC_EVENT_NONE，所以這裡就清除SyncStartEvent；

    2.判斷在mActiveTracks集合中傳過來的recordTrack是否是第一個，而我們這是第一次來，肯定會是第一個，而如果不是第一個，也就是說之前因為某種狀態已經開始了錄音，所以再判斷是否是PAUSING狀態，更新狀態到ACTIVE，然後直接return；

    3.設定recordTrack的狀態為STARTING_1，然後加到mActiveTracks集合中，如果此時再去indexOf的話，肯定就是1了；

    4.判斷recordTrack是否是外部的Track，而isExternalTrack的定義如下：

            bool        isTimedTrack() const { return (mType == TYPE_TIMED); }
            bool        isOutputTrack() const { return (mType == TYPE_OUTPUT); }
            bool        isPatchTrack() const { return (mType == TYPE_PATCH); }
            bool        isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }

再回憶下，我們在new RecordTrack的時候傳入的mType是TrackBase::TYPE_DEFAULT，所以這個recordTrack是外部的Track；

    5.確定是ExternalTrack，那麼就會呼叫AudioSystem::startInput方法開始採集資料，這個sessionId就是上一篇文章中出現的那個了，而對於這個mId，在AudioSystem::startInput中他的型別是audio_io_handle_t，在上一篇文章中，這個io_handle是通過AudioSystem::getInputForAttr獲取到的，獲取到之後通過checkRecordThread_l(input)獲取到了一個RecordThread物件，我們看下RecordThread類：class RecordThread : public ThreadBase，再看下ThreadBase父類，父類的建構函式實現在Threads.cpp檔案中，在這裡我們發現把input賦值給了mId，也就是說，呼叫AudioSystem::startInput函式的引數，就是之前建立的輸入流input以及生成的sessionId了。

AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
        audio_devices_t outDevice, audio_devices_t inDevice, type_t type)
    :   Thread(false /*canCallJava*/),
        mType(type),
        mAudioFlinger(audioFlinger),
        // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize
        // are set by PlaybackThread::readOutputParameters_l() or
        // RecordThread::readInputParameters_l()
        //FIXME: mStandby should be true here. Is this some kind of hack?
        mStandby(false), mOutDevice(outDevice), mInDevice(inDevice),
        mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id),
        // mName will be set by concrete (non-virtual) subclass
        mDeathRecipient(new PMDeathRecipient(this))
{
}

    6.如果mRsmpInRear不為null的話，就重置mRsmpInFront等緩衝區索引；這裡顯然還沒開始錄音，所以mRsmpInRear是null的；

    7.設定recordTrack的狀態為STARTING_2，然後呼叫mWaitWorkCV.broadcast()廣播通知所有的執行緒開始工作。注意：這裡不得不提前劇透下，在AudioSystem::startInput中，AudioFlinger::RecordThread已經開始跑起來了，所以其實broadcast對RecordThread是沒有作用的，並且，需要特別注意的是，這裡更新了recordTrack->mState為STARTING_2，之前在加入mActiveTracks時的狀態是STARTING_1，這個地方比較有意思，這裡先標記下，到時候在分析RecordThread的時候揭曉答案；

    8.判斷下recordTrack是否已經加到mActiveTracks集合中了，如果沒有的話，就說明start失敗了，需要stopInput等；

接下來繼續分析AudioSystem::startInput方法

frameworks\av\media\libmedia\AudioSystem.cpp

status_t AudioSystem::startInput(audio_io_handle_t input,
                                 audio_session_t session)
{
    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
    if (aps == 0) return PERMISSION_DENIED;
    return aps->startInput(input, session);
}

繼續呼叫AudioPolicyService的startInput方法

frameworks\av\services\audiopolicy\AudioPolicyInterfaceImpl.cpp

status_t AudioPolicyService::startInput(audio_io_handle_t input,
                                        audio_session_t session)
{
    if (mAudioPolicyManager == NULL) {
        return NO_INIT;
    }
    Mutex::Autolock _l(mLock);

    return mAudioPolicyManager->startInput(input, session);
}

繼續轉發

frameworks\av\services\audiopolicy\AudioPolicyManager.cpp

status_t AudioPolicyManager::startInput(audio_io_handle_t input,
                                        audio_session_t session)
{
    ssize_t index = mInputs.indexOfKey(input);
    if (index < 0) {
        ALOGW("startInput() unknown input %d", input);
        return BAD_VALUE;
    }
    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);

    index = inputDesc->mSessions.indexOf(session);
    if (index < 0) {
        ALOGW("startInput() unknown session %d on input %d", session, input);
        return BAD_VALUE;
    }

    // virtual input devices are compatible with other input devices
    if (!isVirtualInputDevice(inputDesc->mDevice)) {
        // for a non-virtual input device, check if there is another (non-virtual) active input
        audio_io_handle_t activeInput = getActiveInput();
        if (activeInput != 0 && activeInput != input) {
            // If the already active input uses AUDIO_SOURCE_HOTWORD then it is closed,
            // otherwise the active input continues and the new input cannot be started.
            sp<AudioInputDescriptor> activeDesc = mInputs.valueFor(activeInput);
            if (activeDesc->mInputSource == AUDIO_SOURCE_HOTWORD) {
                ALOGW("startInput(%d) preempting low-priority input %d", input, activeInput);
			
                stopInput(activeInput, activeDesc->mSessions.itemAt(0));
                releaseInput(activeInput, activeDesc->mSessions.itemAt(0));
            } else {
                ALOGE("startInput(%d) failed: other input %d already started", input, activeInput);
                return INVALID_OPERATION;
            }
        }
    }

    if (inputDesc->mRefCount == 0) {
        if (activeInputsCount() == 0) {
            SoundTrigger::setCaptureState(true);
        }
        setInputDevice(input, getNewInputDevice(input), true /* force */);

        // automatically enable the remote submix output when input is started if not
        // used by a policy mix of type MIX_TYPE_RECORDERS
        // For remote submix (a virtual device), we open only one input per capture request.
        if (audio_is_remote_submix_device(inputDesc->mDevice)) {
			ALOGV("audio_is_remote_submix_device(inputDesc->mDevice)");
            String8 address = String8("");
            if (inputDesc->mPolicyMix == NULL) {
                address = String8("0");
            } else if (inputDesc->mPolicyMix->mMixType == MIX_TYPE_PLAYERS) {
                address = inputDesc->mPolicyMix->mRegistrationId;
            }
            if (address != "") {
                setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
                        AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
                        address);
            }
        }
    }

    ALOGV("AudioPolicyManager::startInput() input source = %d", inputDesc->mInputSource);

    inputDesc->mRefCount++;
    return NO_ERROR;
}

在這個函式中主要工作如下：

    1.通過input找到mInputs集合中的位置，並獲取他的inputDesc；

    2.判斷input裝置是否是虛擬裝置，若不是則再判斷是否存在active的裝置，我們第一次來，不存在的！

    3.第一次來嘛，所以會呼叫SoundTrigger::setCaptureState(true)，不過這個是和語音識別有關係，這裡就不多說了；

    4.繼續呼叫setInputDevice函式，其中getNewInputDevice函式的作用是根據input獲取audio_devices_t裝置，同樣，這個裝置在上一篇文章中的AudioPolicyManager::getInputForAttr方法中通過getDeviceAndMixForInputSource獲取到的，即AUDIO_DEVICE_IN_BUILTIN_MIC內建MIC裝置，同時在該函式最後更新了inputDesc->mDevice；

    5.判斷是否是remote_submix裝置，然後做相應處理；

    6.inputDesc的mRefCount計數+1；

繼續分析setInputDevice函式

status_t AudioPolicyManager::setInputDevice(audio_io_handle_t input,
                                            audio_devices_t device,
                                            bool force,
                                            audio_patch_handle_t *patchHandle)
{
    status_t status = NO_ERROR;

    sp<AudioInputDescriptor> inputDesc = mInputs.valueFor(input);
    if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
        inputDesc->mDevice = device;

        DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
        if (!deviceList.isEmpty()) {
            struct audio_patch patch;
            inputDesc->toAudioPortConfig(&patch.sinks[0]);
            // AUDIO_SOURCE_HOTWORD is for internal use only:
            // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
            if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
                    !inputDesc->mIsSoundTrigger) {
                patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
            }
            patch.num_sinks = 1;
            //only one input device for now
            deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
            patch.num_sources = 1;
            ssize_t index;
            if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
                index = mAudioPatches.indexOfKey(*patchHandle);
            } else {
                index = mAudioPatches.indexOfKey(inputDesc->mPatchHandle);
            }
            sp< AudioPatch> patchDesc;
            audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
            if (index >= 0) {
                patchDesc = mAudioPatches.valueAt(index);
                afPatchHandle = patchDesc->mAfPatchHandle;
            }

            status_t status = mpClientInterface->createAudioPatch(&patch,
                                                                  &afPatchHandle,
                                                                  0);
            if (status == NO_ERROR) {
                if (index < 0) {
                    patchDesc = new AudioPatch((audio_patch_handle_t)nextUniqueId(),
                                               &patch, mUidCached);
                    addAudioPatch(patchDesc->mHandle, patchDesc);
                } else {
                    patchDesc->mPatch = patch;
                }
                patchDesc->mAfPatchHandle = afPatchHandle;
                patchDesc->mUid = mUidCached;
                if (patchHandle) {
                    *patchHandle = patchDesc->mHandle;
                }
                inputDesc->mPatchHandle = patchDesc->mHandle;
                nextAudioPortGeneration();
                mpClientInterface->onAudioPatchListUpdate();
            }
        }
    }
    return status;
}

在這個函式中主要的工作如下：

    1.這裡已經知道device與inputDesc->mDevice都已經是AUDIO_DEVICE_IN_BUILTIN_MIC，但是force是true；

    2.通過device獲取mAvailableInputDevices集合中的所有裝置，到此刻，我們還只向該集合中新增一個device；

    3.這裡我們分析下struct audio_patch；他定義在system\core\include\system\audio.h，這裡對audio_patch中的source與sinks進行賦值，注意一點，他把mId（audio_io_handle_t）賦值給了id，然後在這個audio_patch中儲存了InputSource，sample_rate，channel_mask，format，hw_module等等，幾乎都存進去了；

struct audio_patch {
    audio_patch_handle_t id;            /* patch unique ID */
    unsigned int      num_sources;      /* number of sources in following array */
    struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX];
    unsigned int      num_sinks;        /* number of sinks in following array */
    struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX];
};

struct audio_port_config {
    audio_port_handle_t      id;           /* port unique ID */
    audio_port_role_t        role;         /* sink or source */
    audio_port_type_t        type;         /* device, mix ... */
    unsigned int             config_mask;  /* e.g AUDIO_PORT_CONFIG_ALL */
    unsigned int             sample_rate;  /* sampling rate in Hz */
    audio_channel_mask_t     channel_mask; /* channel mask if applicable */
    audio_format_t           format;       /* format if applicable */
    struct audio_gain_config gain;         /* gain to apply if applicable */
    union {
        struct audio_port_config_device_ext  device;  /* device specific info */
        struct audio_port_config_mix_ext     mix;     /* mix specific info */
        struct audio_port_config_session_ext session; /* session specific info */
    } ext;
};
struct audio_port_config_device_ext {
    audio_module_handle_t hw_module;                /* module the device is attached to */
    audio_devices_t       type;                     /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
    char                  address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */
};
struct audio_port_config_mix_ext {
    audio_module_handle_t hw_module;    /* module the stream is attached to */
    audio_io_handle_t handle;           /* I/O handle of the input/output stream */
    union {
        //TODO: change use case for output streams: use strategy and mixer attributes
        audio_stream_type_t stream;
        audio_source_t      source;
    } usecase;
};

    4.呼叫mpClientInterface->createAudioPatch建立Audio通路；

    5.更新patchDesc的屬性；

    6.如果createAudioPatch的status是NO_ERROR的話，就呼叫mpClientInterface->onAudioPatchListUpdate更新AudioPatch列表；

這裡我們著重分析第4、6步：

首先分析下AudioPolicyManager.cpp的AudioPolicyManager::setInputDevice的第4步：建立Audio通路

frameworks\av\services\audiopolicy\AudioPolicyClientImpl.cpp

status_t AudioPolicyService::AudioPolicyClient::createAudioPatch(const struct audio_patch *patch,
                                                                  audio_patch_handle_t *handle,
                                                                  int delayMs)
{
    return mAudioPolicyService->clientCreateAudioPatch(patch, handle, delayMs);
}

繼續向下

frameworks\av\services\audiopolicy\AudioPolicyService.cpp

status_t AudioPolicyService::clientCreateAudioPatch(const struct audio_patch *patch,
                                                audio_patch_handle_t *handle,
                                                int delayMs)
{
    return mAudioCommandThread->createAudioPatchCommand(patch, handle, delayMs);
}

還是在這個檔案中

status_t AudioPolicyService::AudioCommandThread::createAudioPatchCommand(
                                                const struct audio_patch *patch,
                                                audio_patch_handle_t *handle,
                                                int delayMs)
{
    status_t status = NO_ERROR;

    sp<AudioCommand> command = new AudioCommand();
    command->mCommand = CREATE_AUDIO_PATCH;
    CreateAudioPatchData *data = new CreateAudioPatchData();
    data->mPatch = *patch;
    data->mHandle = *handle;
    command->mParam = data;
    command->mWaitStatus = true;
    ALOGV("AudioCommandThread() adding create patch delay %d", delayMs);
    status = sendCommand(command, delayMs);
    if (status == NO_ERROR) {
        *handle = data->mHandle;
    }
    return status;
}

後面就是把audio_patch封裝下，然後加入到AudioCommands佇列中去，所以接下來直接看threadLoop中是怎麼處理的

bool AudioPolicyService::AudioCommandThread::threadLoop()
{
    nsecs_t waitTime = INT64_MAX;

    mLock.lock();
    while (!exitPending())
    {
        sp<AudioPolicyService> svc;
        while (!mAudioCommands.isEmpty() && !exitPending()) {
            nsecs_t curTime = systemTime();
            // commands are sorted by increasing time stamp: execute them from index 0 and up
            if (mAudioCommands[0]->mTime <= curTime) {
                sp<AudioCommand> command = mAudioCommands[0];
                mAudioCommands.removeAt(0);
                mLastCommand = command;

                switch (command->mCommand) {
                case START_TONE: {
                    mLock.unlock();
                    ToneData *data = (ToneData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing start tone %d on stream %d",
                            data->mType, data->mStream);
                    delete mpToneGenerator;
                    mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
                    mpToneGenerator->startTone(data->mType);
                    mLock.lock();
                    }break;
                case STOP_TONE: {
                    mLock.unlock();
                    ALOGV("AudioCommandThread() processing stop tone");
                    if (mpToneGenerator != NULL) {
                        mpToneGenerator->stopTone();
                        delete mpToneGenerator;
                        mpToneGenerator = NULL;
                    }
                    mLock.lock();
                    }break;
                case SET_VOLUME: {
                    VolumeData *data = (VolumeData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing set volume stream %d, \
                            volume %f, output %d", data->mStream, data->mVolume, data->mIO);
                    command->mStatus = AudioSystem::setStreamVolume(data->mStream,
                                                                    data->mVolume,
                                                                    data->mIO);
                    }break;
                case SET_PARAMETERS: {
                    ParametersData *data = (ParametersData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
                            data->mKeyValuePairs.string(), data->mIO);
                    command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
                    }break;
                case SET_VOICE_VOLUME: {
                    VoiceVolumeData *data = (VoiceVolumeData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing set voice volume volume %f",
                            data->mVolume);
                    command->mStatus = AudioSystem::setVoiceVolume(data->mVolume);
                    }break;
                case STOP_OUTPUT: {
                    StopOutputData *data = (StopOutputData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing stop output %d",
                            data->mIO);
                    svc = mService.promote();
                    if (svc == 0) {
                        break;
                    }
                    mLock.unlock();
                    svc->doStopOutput(data->mIO, data->mStream, data->mSession);
                    mLock.lock();
                    }break;
                case RELEASE_OUTPUT: {
                    ReleaseOutputData *data = (ReleaseOutputData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing release output %d",
                            data->mIO);
                    svc = mService.promote();
                    if (svc == 0) {
                        break;
                    }
                    mLock.unlock();
                    svc->doReleaseOutput(data->mIO, data->mStream, data->mSession);
                    mLock.lock();
                    }break;
                case CREATE_AUDIO_PATCH: {
                    CreateAudioPatchData *data = (CreateAudioPatchData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing create audio patch");
                    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
                    if (af == 0) {
                        command->mStatus = PERMISSION_DENIED;
                    } else {
                        command->mStatus = af->createAudioPatch(&data->mPatch, &data->mHandle);
                    }
                    } break;
                case RELEASE_AUDIO_PATCH: {
                    ReleaseAudioPatchData *data = (ReleaseAudioPatchData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing release audio patch");
                    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
                    if (af == 0) {
                        command->mStatus = PERMISSION_DENIED;
                    } else {
                        command->mStatus = af->releaseAudioPatch(data->mHandle);
                    }
                    } break;
                case UPDATE_AUDIOPORT_LIST: {
                    ALOGV("AudioCommandThread() processing update audio port list");
                    svc = mService.promote();
                    if (svc == 0) {
                        break;
                    }
                    mLock.unlock();
                    svc->doOnAudioPortListUpdate();
                    mLock.lock();
                    }break;
                case UPDATE_AUDIOPATCH_LIST: {
                    ALOGV("AudioCommandThread() processing update audio patch list");
                    svc = mService.promote();
                    if (svc == 0) {
                        break;
                    }
                    mLock.unlock();
                    svc->doOnAudioPatchListUpdate();
                    mLock.lock();
                    }break;
                case SET_AUDIOPORT_CONFIG: {
                    SetAudioPortConfigData *data = (SetAudioPortConfigData *)command->mParam.get();
                    ALOGV("AudioCommandThread() processing set port config");
                    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
                    if (af == 0) {
                        command->mStatus = PERMISSION_DENIED;
                    } else {
                        command->mStatus = af->setAudioPortConfig(&data->mConfig);
                    }
                    } break;
                default:
                    ALOGW("AudioCommandThread() unknown command %d", command->mCommand);
                }
                {
                    Mutex::Autolock _l(command->mLock);
                    if (command->mWaitStatus) {
                        command->mWaitStatus = false;
                        command->mCond.signal();
                    }
                }
                waitTime = INT64_MAX;
            } else {
                waitTime = mAudioCommands[0]->mTime - curTime;
                break;
            }
        }
        // release mLock before releasing strong reference on the service as
        // AudioPolicyService destructor calls AudioCommandThread::exit() which acquires mLock.
        mLock.unlock();
        svc.clear();
        mLock.lock();
        if (!exitPending() && mAudioCommands.isEmpty()) {
            // release delayed commands wake lock
            release_wake_lock(mName.string());
            ALOGV("AudioCommandThread() going to sleep");
            mWaitWorkCV.waitRelative(mLock, waitTime);
            ALOGV("AudioCommandThread() waking up");
        }
    }
    // release delayed commands wake lock before quitting
    if (!mAudioCommands.isEmpty()) {
        release_wake_lock(mName.string());
    }
    mLock.unlock();
    return false;
}

這裡直接看CREATE_AUDIO_PATCH的分支，他呼叫了AF端的af->createAudioPatch函式，同樣在這個loop中，也有後面的UPDATE_AUDIOPATCH_LIST分支

frameworks\av\services\audioflinger\PatchPanel.cpp

status_t AudioFlinger::createAudioPatch(const struct audio_patch *patch,
                                   audio_patch_handle_t *handle)
{
    Mutex::Autolock _l(mLock);
    if (mPatchPanel != 0) {
        return mPatchPanel->createAudioPatch(patch, handle);
    }
    return NO_INIT;
}

繼續往下走 （唉，老實說我都不想走了，繞來繞去的。。

status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,
                                   audio_patch_handle_t *handle)
{
    ALOGV("createAudioPatch() num_sources %d num_sinks %d handle %d",
          patch->num_sources, patch->num_sinks, *handle);
    status_t status = NO_ERROR;
    audio_patch_handle_t halHandle = AUDIO_PATCH_HANDLE_NONE;
    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
    if (audioflinger == 0) {
        return NO_INIT;
    }

    if (handle == NULL || patch == NULL) {
        return BAD_VALUE;
    }
    if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
            patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
        return BAD_VALUE;
    }
    // limit number of sources to 1 for now or 2 sources for special cross hw module case.
    // only the audio policy manager can request a patch creation with 2 sources.
    if (patch->num_sources > 2) {
        return INVALID_OPERATION;
    }

    if (*handle != AUDIO_PATCH_HANDLE_NONE) {
        for (size_t index = 0; *handle != 0 && index < mPatches.size(); index++) {
            if (*handle == mPatches[index]->mHandle) {
                ALOGV("createAudioPatch() removing patch handle %d", *handle);
                halHandle = mPatches[index]->mHalHandle;
                Patch *removedPatch = mPatches[index];
                mPatches.removeAt(index);
                delete removedPatch;
                break;
            }
        }
    }

    Patch *newPatch = new Patch(patch);

    switch (patch->sources[0].type) {
        case AUDIO_PORT_TYPE_DEVICE: {
            audio_module_handle_t srcModule = patch->sources[0].ext.device.hw_module;
            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
            if (index < 0) {
                ALOGW("createAudioPatch() bad src hw module %d", srcModule);
                status = BAD_VALUE;
                goto exit;
            }
            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
            for (unsigned int i = 0; i < patch->num_sinks; i++) {
                // support only one sink if connection to a mix or across HW modules
                if ((patch->sinks[i].type == AUDIO_PORT_TYPE_MIX ||
                        patch->sinks[i].ext.mix.hw_module != srcModule) &&
                        patch->num_sinks > 1) {
                    status = INVALID_OPERATION;
                    goto exit;
                }
                // reject connection to different sink types
                if (patch->sinks[i].type != patch->sinks[0].type) {
                    ALOGW("createAudioPatch() different sink types in same patch not supported");
                    status = BAD_VALUE;
                    goto exit;
                }
                // limit to connections between devices and input streams for HAL before 3.0
                if (patch->sinks[i].ext.mix.hw_module == srcModule &&
                        (audioHwDevice->version() < AUDIO_DEVICE_API_VERSION_3_0) &&
                        (patch->sinks[i].type != AUDIO_PORT_TYPE_MIX)) {
                    ALOGW("createAudioPatch() invalid sink type %d for device source",
                          patch->sinks[i].type);
                    status = BAD_VALUE;
                    goto exit;
                }
            }

            if (patch->sinks[0].ext.device.hw_module != srcModule) {
                // limit to device to device connection if not on same hw module
                if (patch->sinks[0].type != AUDIO_PORT_TYPE_DEVICE) {
                    ALOGW("createAudioPatch() invalid sink type for cross hw module");
                    status = INVALID_OPERATION;
                    goto exit;
                }
                // special case num sources == 2 -=> reuse an exiting output mix to connect to the
                // sink
                if (patch->num_sources == 2) {
                    if (patch->sources[1].type != AUDIO_PORT_TYPE_MIX ||
                            patch->sinks[0].ext.device.hw_module !=
                                    patch->sources[1].ext.mix.hw_module) {
                        ALOGW("createAudioPatch() invalid source combination");
                        status = INVALID_OPERATION;
                        goto exit;
                    }

                    sp<ThreadBase> thread =
                            audioflinger->checkPlaybackThread_l(patch->sources[1].ext.mix.handle);
                    newPatch->mPlaybackThread = (MixerThread *)thread.get();
                    if (thread == 0) {
                        ALOGW("createAudioPatch() cannot get playback thread");
                        status = INVALID_OPERATION;
                        goto exit;
                    }
                } else {
                    audio_config_t config = AUDIO_CONFIG_INITIALIZER;
                    audio_devices_t device = patch->sinks[0].ext.device.type;
                    String8 address = String8(patch->sinks[0].ext.device.address);
                    audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
                    newPatch->mPlaybackThread = audioflinger->openOutput_l(
                                                             patch->sinks[0].ext.device.hw_module,
                                                             &output,
                                                             &config,
                                                             device,
                                                             address,
                                                             AUDIO_OUTPUT_FLAG_NONE);
                    ALOGV("audioflinger->openOutput_l() returned %p",
                                          newPatch->mPlaybackThread.get());
                    if (newPatch->mPlaybackThread == 0) {
                        status = NO_MEMORY;
                        goto exit;
                    }
                }
                uint32_t channelCount = newPatch->mPlaybackThread->channelCount();
                audio_devices_t device = patch->sources[0].ext.device.type;
                String8 address = String8(patch->sources[0].ext.device.address);
                audio_config_t config = AUDIO_CONFIG_INITIALIZER;
                audio_channel_mask_t inChannelMask = audio_channel_in_mask_from_count(channelCount);
                config.sample_rate = newPatch->mPlaybackThread->sampleRate();
                config.channel_mask = inChannelMask;
                config.format = newPatch->mPlaybackThread->format();
                audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
                newPatch->mRecordThread = audioflinger->openInput_l(srcModule,
                                                                    &input,
                                                                    &config,
                                                                    device,
                                                                    address,
                                                                    AUDIO_SOURCE_MIC,
                                                                    AUDIO_INPUT_FLAG_NONE);
                ALOGV("audioflinger->openInput_l() returned %p inChannelMask %08x",
                      newPatch->mRecordThread.get(), inChannelMask);
                if (newPatch->mRecordThread == 0) {
                    status = NO_MEMORY;
                    goto exit;
                }
                status = createPatchConnections(newPatch, patch);
                if (status != NO_ERROR) {
                    goto exit;
                }
            } else {
                if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
                    if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
                        sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
                                                                  patch->sinks[0].ext.mix.handle);
                        if (thread == 0) {
                            ALOGW("createAudioPatch() bad capture I/O handle %d",
                                                                  patch->sinks[0].ext.mix.handle);
                            status = BAD_VALUE;
                            goto exit;
                        }
                        status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
                    } else {
                        audio_hw_device_t *hwDevice = audioHwDevice->hwDevice();
                        status = hwDevice->create_audio_patch(hwDevice,
                                                               patch->num_sources,
                                                               patch->sources,
                                                               patch->num_sinks,
                                                               patch->sinks,
                                                               &halHandle);
                    }
                } else {
                    sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
                                                                    patch->sinks[0].ext.mix.handle);
                    if (thread == 0) {
                        ALOGW("createAudioPatch() bad capture I/O handle %d",
                                                                    patch->sinks[0].ext.mix.handle);
                        status = BAD_VALUE;
                        goto exit;
                    }
                    char *address;
                    if (strcmp(patch->sources[0].ext.device.address, "") != 0) {
                        address = audio_device_address_to_parameter(
                                                            patch->sources[0].ext.device.type,
                                                            patch->sources[0].ext.device.address);
                    } else {
                        address = (char *)calloc(1, 1);
                    }
                    AudioParameter param = AudioParameter(String8(address));
                    free(address);
                    param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING),
                                 (int)patch->sources[0].ext.device.type);
                    param.addInt(String8(AUDIO_PARAMETER_STREAM_INPUT_SOURCE),
                                                     (int)patch->sinks[0].ext.mix.usecase.source);
                    ALOGV("createAudioPatch() AUDIO_PORT_TYPE_DEVICE setParameters %s",
                                                                      param.toString().string());
                    status = thread->setParameters(param.toString());
                }
            }
        } break;
        case AUDIO_PORT_TYPE_MIX: {
            audio_module_handle_t srcModule =  patch->sources[0].ext.mix.hw_module;
            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
            if (index < 0) {
                ALOGW("createAudioPatch() bad src hw module %d", srcModule);
                status = BAD_VALUE;
                goto exit;
            }
            // limit to connections between devices and output streams
            for (unsigned int i = 0; i < patch->num_sinks; i++) {
                if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
                    ALOGW("createAudioPatch() invalid sink type %d for mix source",
                          patch->sinks[i].type);
                    status = BAD_VALUE;
                    goto exit;
                }
                // limit to connections between sinks and sources on same HW module
                if (patch->sinks[i].ext.device.hw_module != srcModule) {
                    status = BAD_VALUE;
                    goto exit;
                }
            }
            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
            sp<ThreadBase> thread =
                            audioflinger->checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
            if (thread == 0) {
                ALOGW("createAudioPatch() bad playback I/O handle %d",
                          patch->sources[0].ext.mix.handle);
                status = BAD_VALUE;
                goto exit;
            }
            if (audioHwDevice->version() >= AUDIO_DEVICE_API_VERSION_3_0) {
                status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
            } else {
                audio_devices_t type = AUDIO_DEVICE_NONE;
                for (unsigned int i = 0; i < patch->num_sinks; i++) {
                    type |= patch->sinks[i].ext.device.type;
                }
                char *address;
                if (strcmp(patch->sinks[0].ext.device.address, "") != 0) {
                    //FIXME: we only support address on first sink with HAL version < 3.0
                    address = audio_device_address_to_parameter(
                                                                patch->sinks[0].ext.device.type,
                                                                patch->sinks[0].ext.device.address);
                } else {
                    address = (char *)calloc(1, 1);
                }
                AudioParameter param = AudioParameter(String8(address));
                free(address);
                param.addInt(String8(AUDIO_PARAMETER_STREAM_ROUTING), (int)type);
                status = thread->setParameters(param.toString());
            }

        } break;
        default:
            status = BAD_VALUE;
            goto exit;
    }
exit:
    ALOGV("createAudioPatch() status %d", status);
    if (status == NO_ERROR) {
        *handle = audioflinger->nextUniqueId();
        newPatch->mHandle = *handle;
        newPatch->mHalHandle = halHandle;
        mPatches.add(newPatch);
        ALOGV("createAudioPatch() added new patch handle %d halHandle %d", *handle, halHandle);
    } else {
        clearPatchConnections(newPatch);
        delete newPatch;
    }
    return status;
}

在這個函式中主要工作如下：

    1.在AudioPolicyManager::setInputDevice()函式中，num_sources與num_sinks都為1；

    2.當halHandle不是AUDIO_PATCH_HANDLE_NONE的時候，就去mPatches集合中找到這個halHandle，然後刪除他，而在這裡，halHandle就是AUDIO_PATCH_HANDLE_NONE；

    3.這裡的source.type為AUDIO_PORT_TYPE_DEVICE，獲取patch中的audio_module_handle_t，獲取AF端的AudioHwDevice，後面有個for迴圈判斷，根據之前的引數設定，均不會進到if裡面；

    4.判斷source裡的audio_module_handle_t與sink裡的是否一致，那肯定一致噻；

    5.再判斷hal程式碼中的version版本，我們看下hardware\aw\audio\tulip\audio_hw.c的adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0；

    6.呼叫AF端的checkRecordThread_l函式，即通過audio_io_handle_t從mRecordThreads中獲取到RecordThread執行緒；

    7.通過address建立一個AudioParameter物件，並把source.type與source放入AudioParameter物件中；

    8.呼叫thread->setParameters把AudioParameter物件傳遞過去

這裡我們繼續分析下第8步：thread->setParameters

frameworks\av\services\audioflinger\Threads.cpp

status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
    status_t status;

    Mutex::Autolock _l(mLock);

    return sendSetParameterConfigEvent_l(keyValuePairs);
}

emmm，感覺又要繞上一大圈

status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair)
{
    sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair);
    return sendConfigEvent_l(configEvent);
}

把AudioParameter物件轉化為ConfigEvent物件，繼續呼叫

status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event)
{
    status_t status = NO_ERROR;

    mConfigEvents.add(event);
    mWaitWorkCV.signal();
    mLock.unlock();
    {
        Mutex::Autolock _l(event->mLock);
        while (event->mWaitStatus) {
            if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) {
                event->mStatus = TIMED_OUT;
                event->mWaitStatus = false;
            }
        }
        status = event->mStatus;
    }
    mLock.lock();
    return status;
}

把ConfigEvent加入到mConfigEvents中，然後呼叫mWaitWorkCV.signal()通知RecordThread執行緒可以start了，線上程中接受到mWaitWorkCV.wait(mLock);時就會回到reacquire_wakelock位置，再繼續往下，這時候呼叫了processConfigEvents_l，他就是用來處理ConfigEvent事件的

void AudioFlinger::ThreadBase::processConfigEvents_l()
{
    bool configChanged = false;

    while (!mConfigEvents.isEmpty()) {
        ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size());
        sp<ConfigEvent> event = mConfigEvents[0];
        mConfigEvents.removeAt(0);
        switch (event->mType) {
        case CFG_EVENT_PRIO: {
            PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get();
            // FIXME Need to understand why this has to be done asynchronously
            int err = requestPriority(data->mPid, data->mTid, data->mPrio,
                    true /*asynchronous*/);
            if (err != 0) {
                ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
                      data->mPrio, data->mPid, data->mTid, err);
            }
        } break;
        case CFG_EVENT_IO: {
            IoConfigEventData *data = (IoConfigEventData *)event->mData.get();
            audioConfigChanged(data->mEvent, data->mParam);
        } break;
        case CFG_EVENT_SET_PARAMETER: {
            SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get();
            if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) {
                configChanged = true;
            }
        } break;
        case CFG_EVENT_CREATE_AUDIO_PATCH: {
            CreateAudioPatchConfigEventData *data =
                                            (CreateAudioPatchConfigEventData *)event->mData.get();
            event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle);
        } break;
        case CFG_EVENT_RELEASE_AUDIO_PATCH: {
            ReleaseAudioPatchConfigEventData *data =
                                            (ReleaseAudioPatchConfigEventData *)event->mData.get();
            event->mStatus = releaseAudioPatch_l(data->mHandle);
        } break;
        default:
            ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
            break;
        }
        {
            Mutex::Autolock _l(event->mLock);
            if (event->mWaitStatus) {
                event->mWaitStatus = false;
                event->mCond.signal();
            }
        }
        ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this);
    }

    if (configChanged) {
        cacheParameters_l();
    }
}

這裡果然是一直在等待處理mConfigEvents中的事件，而這個event->mType是CFG_EVENT_SET_PARAMETER，所以繼續呼叫checkForNewParameter_l函式，而他肯定是呼叫的是RecordThread中的啦，這..繞了真·一大圈。

bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair,
                                                        status_t& status)
{
    bool reconfig = false;

    status = NO_ERROR;

    audio_format_t reqFormat = mFormat;
    uint32_t samplingRate = mSampleRate;
    audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount);

    AudioParameter param = AudioParameter(keyValuePair);
    int value;

    if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) {
        samplingRate = value;
        reconfig = true;
    }
    if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
        if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
            status = BAD_VALUE;
        } else {
            reqFormat = (audio_format_t) value;
            reconfig = true;
        }
    }
    if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
        audio_channel_mask_t mask = (audio_channel_mask_t) value;
        if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) {
            status = BAD_VALUE;
        } else {
            channelMask = mask;
            reconfig = true;
        }
    }
    if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
        // do not accept frame count changes if tracks are open as the track buffer
        // size depends on frame count and correct behavior would not be guaranteed
        // if frame count is changed after track creation
        if (mActiveTracks.size() > 0) {
            status = INVALID_OPERATION;
        } else {
            reconfig = true;
        }
    }
    if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) {
        // forward device change to effects that have requested to be
        // aware of attached audio device.
        for (size_t i = 0; i < mEffectChains.size(); i++) {
            mEffectChains[i]->setDevice_l(value);
        }