AudioRecord源码解读(3)

    public AudioRecord(int audioSource, int sampleRateInHz, int channelConfig, int audioFormat,
            int bufferSizeInBytes)

       AudioSource.DEFAULT,
        AudioSource.MIC,
        AudioSource.VOICE_UPLINK,
        AudioSource.VOICE_DOWNLINK,
        AudioSource.VOICE_CALL,
        AudioSource.CAMCORDER,
        AudioSource.VOICE_RECOGNITION,
        AudioSource.VOICE_COMMUNICATION,
        AudioSource.UNPROCESSED,
        AudioSource.VOICE_PERFORMANCE,

    static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
        int channelCount = 0;
        switch (channelConfig) {
        case AudioFormat.CHANNEL_IN_DEFAULT: // AudioFormat.CHANNEL_CONFIGURATION_DEFAULT
        case AudioFormat.CHANNEL_IN_MONO:
        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
            channelCount = 1;
            break;
        case AudioFormat.CHANNEL_IN_STEREO:
        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
        case (AudioFormat.CHANNEL_IN_FRONT | AudioFormat.CHANNEL_IN_BACK):
            channelCount = 2;
            break;
        case AudioFormat.CHANNEL_INVALID:
        default:
            loge("getMinBufferSize(): Invalid channel configuration.");
            return ERROR_BAD_VALUE;
        }

        int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
        if (size == 0) {
            return ERROR_BAD_VALUE;
        }
        else if (size == -1) {
            return ERROR;
        }
        else {
            return size;
        }
    }

static jint android_media_AudioRecord_get_min_buff_size(JNIEnv *env,  jobject thiz,
    jint sampleRateInHertz, jint channelCount, jint audioFormat) {
    size_t frameCount = 0;
    audio_format_t format = audioFormatToNative(audioFormat); // 把java 的格式转成native的
    status_t result = AudioRecord::getMinFrameCount(&frameCount,  // 获取最小帧数
            sampleRateInHertz,
            format,
            audio_channel_in_mask_from_count(channelCount));
    return frameCount * channelCount * audio_bytes_per_sample(format);
}

status_t AudioRecord::getMinFrameCount(
        size_t* frameCount,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask)
{
    if (frameCount == NULL) {
        return BAD_VALUE;
    }

    size_t size;
    status_t status = AudioSystem::getInputBufferSize(sampleRate, format, channelMask, &size);
    if (status != NO_ERROR) {
        ALOGE("%s(): AudioSystem could not query the input buffer size for"
              " sampleRate %u, format %#x, channelMask %#x; status %d",
               __func__, sampleRate, format, channelMask, status);
        return status;
    }

    // We double the size of input buffer for ping pong use of record buffer.
    // Assumes audio_is_linear_pcm(format)
    if ((*frameCount = (size * 2) / (audio_channel_count_from_in_mask(channelMask) *
            audio_bytes_per_sample(format))) == 0) {
        ALOGE("%s(): Unsupported configuration: sampleRate %u, format %#x, channelMask %#x",
                __func__, sampleRate, format, channelMask);
        return BAD_VALUE;
    }

    return NO_ERROR;
}

status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
        audio_channel_mask_t channelMask, size_t* buffSize)
{
    const sp<AudioFlingerClient> afc = getAudioFlingerClient();
    if (afc == 0) {
        return NO_INIT;
    }
    return afc->getInputBufferSize(sampleRate, format, channelMask, buffSize);
}

status_t AudioSystem::AudioFlingerClient::getInputBufferSize(
                                                uint32_t sampleRate, audio_format_t format,
                                                audio_channel_mask_t channelMask, size_t* buffSize)
{
    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger(); 
    if (af == 0) {
        return PERMISSION_DENIED;
    }
    Mutex::Autolock _l(mLock);
    // Do we have a stale mInBuffSize or are we requesting the input buffer size for new values
    if ((mInBuffSize == 0) || (sampleRate != mInSamplingRate) || (format != mInFormat)
        || (channelMask != mInChannelMask)) {
        size_t inBuffSize = af->getInputBufferSize(sampleRate, format, channelMask);
        if (inBuffSize == 0) {
            ALOGE("AudioSystem::getInputBufferSize failed sampleRate %d format %#x channelMask %#x",
                    sampleRate, format, channelMask);
            return BAD_VALUE;
        }
        // A benign race is possible here: we could overwrite a fresher cache entry
        // save the request params
        mInSamplingRate = sampleRate;
        mInFormat = format;
        mInChannelMask = channelMask;

        mInBuffSize = inBuffSize;
    }

    *buffSize = mInBuffSize;

    return NO_ERROR;
}

size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format,
        audio_channel_mask_t channelMask) const
{
    status_t ret = initCheck();

    AutoMutex lock(mHardwareLock);

    mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;

    sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice();
    std::vector<audio_channel_mask_t> channelMasks = {channelMask};
    if (channelMask != AUDIO_CHANNEL_IN_MONO)
        channelMasks.push_back(AUDIO_CHANNEL_IN_MONO);
    if (channelMask != AUDIO_CHANNEL_IN_STEREO)
        channelMasks.push_back(AUDIO_CHANNEL_IN_STEREO);

    std::vector<audio_format_t> formats = {format};
    if (format != AUDIO_FORMAT_PCM_16_BIT)
        formats.push_back(AUDIO_FORMAT_PCM_16_BIT);

    std::vector<uint32_t> sampleRates = {sampleRate};
    static const uint32_t SR_44100 = 44100;
    static const uint32_t SR_48000 = 48000;

    if (sampleRate != SR_48000)
        sampleRates.push_back(SR_48000);
    if (sampleRate != SR_44100)
        sampleRates.push_back(SR_44100);

    mHardwareStatus = AUDIO_HW_IDLE;

    // Change parameters of the configuration each iteration until we find a
    // configuration that the device will support.
    audio_config_t config = AUDIO_CONFIG_INITIALIZER;
    for (auto testChannelMask : channelMasks) {
        config.channel_mask = testChannelMask;
        for (auto testFormat : formats) {
            config.format = testFormat;
            for (auto testSampleRate : sampleRates) {
                config.sample_rate = testSampleRate;

                size_t bytes = 0;
                status_t result = dev->getInputBufferSize(&config, &bytes); // 从hal层获取bytes
                if (result != OK || bytes == 0) {
                    continue;
                }

                if (config.sample_rate != sampleRate || config.channel_mask != channelMask ||
                    config.format != format) {
                    uint32_t dstChannelCount = audio_channel_count_from_in_mask(channelMask);
                    uint32_t srcChannelCount =
                        audio_channel_count_from_in_mask(config.channel_mask);
                    size_t srcFrames =
                        bytes / audio_bytes_per_frame(srcChannelCount, config.format);
                    size_t dstFrames = destinationFramesPossible(
                        srcFrames, config.sample_rate, sampleRate);
                    bytes = dstFrames * audio_bytes_per_frame(dstChannelCount, format); // 重新计算bytes
                }
                return bytes;
            }
        }
    }
    return 0;
}

   public AudioRecord(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
            int sessionId) throws IllegalArgumentException {
        mRecordingState = RECORDSTATE_STOPPED;

        // remember which looper is associated with the AudioRecord instanciation
        if ((mInitializationLooper = Looper.myLooper()) == null) {
            mInitializationLooper = Looper.getMainLooper();
        }

        // is this AudioRecord using REMOTE_SUBMIX at full volume?
        if (attributes.getCapturePreset() == MediaRecorder.AudioSource.REMOTE_SUBMIX) {
            final AudioAttributes.Builder filteredAttr = new AudioAttributes.Builder();
            final Iterator<String> tagsIter = attributes.getTags().iterator();
            while (tagsIter.hasNext()) {
                final String tag = tagsIter.next();
                if (tag.equalsIgnoreCase(SUBMIX_FIXED_VOLUME)) {
                    mIsSubmixFullVolume = true;
                    Log.v(TAG, "Will record from REMOTE_SUBMIX at full fixed volume");
                } else { // SUBMIX_FIXED_VOLUME: is not to be propagated to the native layers
                    filteredAttr.addTag(tag);
                }
            }
            filteredAttr.setInternalCapturePreset(attributes.getCapturePreset());
            mAudioAttributes = filteredAttr.build();
        } else {
            mAudioAttributes = attributes;
        }

        int rate = format.getSampleRate();
        if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
            rate = 0;
        }

        int encoding = AudioFormat.ENCODING_DEFAULT;
        if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0)
        {
            encoding = format.getEncoding();
        }

        audioParamCheck(attributes.getCapturePreset(), rate, encoding);

        if ((format.getPropertySetMask()
                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
            mChannelIndexMask = format.getChannelIndexMask();
            mChannelCount = format.getChannelCount();
        }
        if ((format.getPropertySetMask()
                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
            mChannelMask = getChannelMaskFromLegacyConfig(format.getChannelMask(), false);
            mChannelCount = format.getChannelCount();
        } else if (mChannelIndexMask == 0) {
            mChannelMask = getChannelMaskFromLegacyConfig(AudioFormat.CHANNEL_IN_DEFAULT, false);
            mChannelCount =  AudioFormat.channelCountFromInChannelMask(mChannelMask);
        }

        audioBuffSizeCheck(bufferSizeInBytes);

        int[] sampleRate = new int[] {mSampleRate};
        int[] session = new int[1];
        session[0] = sessionId;
        //TODO: update native initialization when information about hardware init failure
        //      due to capture device already open is available.
        int initResult = native_setup( new WeakReference<AudioRecord>(this),
                mAudioAttributes, sampleRate, mChannelMask, mChannelIndexMask,
                mAudioFormat, mNativeBufferSizeInBytes,
                session, getCurrentOpPackageName(), 0 /*nativeRecordInJavaObj*/); // 真正实现的地方
        if (initResult != SUCCESS) {
            loge("Error code "+initResult+" when initializing native AudioRecord object.");
            return; // with mState == STATE_UNINITIALIZED
        }

        mSampleRate = sampleRate[0];
        mSessionId = session[0];

        mState = STATE_INITIALIZED;
    }

static jint
android_media_AudioRecord_setup(JNIEnv *env, jobject thiz, jobject weak_this,
        jobject jaa, jintArray jSampleRate, jint channelMask, jint channelIndexMask,
        jint audioFormat, jint buffSizeInBytes, jintArray jSession, jstring opPackageName,
        jlong nativeRecordInJavaObj)
{
    audio_channel_mask_t localChanMask = inChannelMaskToNative(channelMask);
    jint* nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
  
    audio_session_t sessionId = (audio_session_t) nSession[0];
    env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
    nSession = NULL;

    sp<AudioRecord> lpRecorder = 0;
    audiorecord_callback_cookie *lpCallbackData = NULL;

    jclass clazz = env->GetObjectClass(thiz);
    if (clazz == NULL) {
        ALOGE("Can't find %s when setting up callback.", kClassPathName);
        return (jint) AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED;
    }

    // if we pass in an existing *Native* AudioRecord, we don't need to create/initialize one.
    if (nativeRecordInJavaObj == 0) {  // native的audio record还没创建，那么就需要创建
    
        jint elements[1];
        env->GetIntArrayRegion(jSampleRate, 0, 1, elements);
        int sampleRateInHertz = elements[0];

        // channel index mask takes priority over channel position masks.
        if (channelIndexMask) {
            // Java channel index masks need the representation bits set.
            localChanMask = audio_channel_mask_from_representation_and_bits(
                    AUDIO_CHANNEL_REPRESENTATION_INDEX,
                    channelIndexMask);
        }
        // Java channel position masks map directly to the native definition

        if (!audio_is_input_channel(localChanMask)) {
            ALOGE("Error creating AudioRecord: channel mask %#x is not valid.", localChanMask);
            return (jint) AUDIORECORD_ERROR_SETUP_INVALIDCHANNELMASK;
        }
        uint32_t channelCount = audio_channel_count_from_in_mask(localChanMask);

        // compare the format against the Java constants
        audio_format_t format = audioFormatToNative(audioFormat);
        if (format == AUDIO_FORMAT_INVALID) {
            ALOGE("Error creating AudioRecord: unsupported audio format %d.", audioFormat);
            return (jint) AUDIORECORD_ERROR_SETUP_INVALIDFORMAT;
        }

        size_t bytesPerSample = audio_bytes_per_sample(format);

        if (buffSizeInBytes == 0) {
             ALOGE("Error creating AudioRecord: frameCount is 0.");
            return (jint) AUDIORECORD_ERROR_SETUP_ZEROFRAMECOUNT;
        }
        size_t frameSize = channelCount * bytesPerSample;
        size_t frameCount = buffSizeInBytes / frameSize;

        ScopedUtfChars opPackageNameStr(env, opPackageName);

        // create an uninitialized AudioRecord object
        lpRecorder = new AudioRecord(String16(opPackageNameStr.c_str())); // 创建AudioRecord

        // read the AudioAttributes values
        auto paa = JNIAudioAttributeHelper::makeUnique();
        jint jStatus = JNIAudioAttributeHelper::nativeFromJava(env, jaa, paa.get());
        if (jStatus != (jint)AUDIO_JAVA_SUCCESS) {
            return jStatus;
        }
        ALOGV("AudioRecord_setup for source=%d tags=%s flags=%08x", paa->source, paa->tags, paa->flags);

        audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE;
        if (paa->flags & AUDIO_FLAG_HW_HOTWORD) {
            flags = AUDIO_INPUT_FLAG_HW_HOTWORD;
        }
        // create the callback information:
        // this data will be passed with every AudioRecord callback
        lpCallbackData = new audiorecord_callback_cookie;
        lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);
        // we use a weak reference so the AudioRecord object can be garbage collected.
        lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);
        lpCallbackData->busy = false;

        const status_t status = lpRecorder->set(paa->source, // set 配置
            sampleRateInHertz,
            format,        // word length, PCM
            localChanMask,
            frameCount,
            recorderCallback,// callback_t
            lpCallbackData,// void* user
            0,             // notificationFrames,
            true,          // threadCanCallJava
            sessionId,
            AudioRecord::TRANSFER_DEFAULT,
            flags,
            -1, -1,        // default uid, pid
            paa.get());

        if (status != NO_ERROR) {
            ALOGE("Error creating AudioRecord instance: initialization check failed with status %d.",
                    status);
            goto native_init_failure;
        }
        // Set caller name so it can be logged in destructor.
        // MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_JAVA
        lpRecorder->setCallerName("java");
    } else { // end if nativeRecordInJavaObj == 0) // 已经创建了native的AudioRecord
        lpRecorder = (AudioRecord*)nativeRecordInJavaObj;

        // create the callback information:
        // this data will be passed with every AudioRecord callback
        lpCallbackData = new audiorecord_callback_cookie;
        lpCallbackData->audioRecord_class = (jclass)env->NewGlobalRef(clazz);
        // we use a weak reference so the AudioRecord object can be garbage collected.
        lpCallbackData->audioRecord_ref = env->NewGlobalRef(weak_this);
        lpCallbackData->busy = false;
    }

    nSession = (jint *) env->GetPrimitiveArrayCritical(jSession, NULL);
    if (nSession == NULL) {
        ALOGE("Error creating AudioRecord: Error retrieving session id pointer");
        goto native_init_failure;
    }
    // read the audio session ID back from AudioRecord in case a new session was created during set()
    nSession[0] = lpRecorder->getSessionId();
    env->ReleasePrimitiveArrayCritical(jSession, nSession, 0);
    nSession = NULL;

    {
        const jint elements[1] = { (jint) lpRecorder->getSampleRate() };
        env->SetIntArrayRegion(jSampleRate, 0, 1, elements);
    }

    {   // scope for the lock
        Mutex::Autolock l(sLock);
        sAudioRecordCallBackCookies.add(lpCallbackData);
    }
    // save our newly created C++ AudioRecord in the "nativeRecorderInJavaObj" field
    // of the Java object
    setAudioRecord(env, thiz, lpRecorder); // 关联Java对象

    // save our newly created callback information in the "nativeCallbackCookie" field
    // of the Java object (in mNativeCallbackCookie) so we can free the memory in finalize()
    env->SetLongField(thiz, javaAudioRecordFields.nativeCallbackCookie, (jlong)lpCallbackData);

    return (jint) AUDIO_JAVA_SUCCESS;

    // failure:
native_init_failure:
    env->DeleteGlobalRef(lpCallbackData->audioRecord_class);
    env->DeleteGlobalRef(lpCallbackData->audioRecord_ref);
    delete lpCallbackData;
    env->SetLongField(thiz, javaAudioRecordFields.nativeCallbackCookie, 0);

    // lpRecorder goes out of scope, so reference count drops to zero
    return (jint) AUDIORECORD_ERROR_SETUP_NATIVEINITFAILED;
}

AudioRecord::AudioRecord(
        audio_source_t inputSource,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        const String16& opPackageName,
        size_t frameCount,
        callback_t cbf,
        void* user,
        uint32_t notificationFrames,
        audio_session_t sessionId,
        transfer_type transferType,
        audio_input_flags_t flags,
        uid_t uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        audio_port_handle_t selectedDeviceId,
        audio_microphone_direction_t selectedMicDirection,
        float microphoneFieldDimension)
    : mActive(false),
      mStatus(NO_INIT),
      mOpPackageName(opPackageName),
      mSessionId(AUDIO_SESSION_ALLOCATE),
      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
      mPreviousSchedulingGroup(SP_DEFAULT),
      mProxy(NULL)
{
    (void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
            notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
            uid, pid, pAttributes, selectedDeviceId,
            selectedMicDirection, microphoneFieldDimension);
}

status_t AudioRecord::set(
        audio_source_t inputSource,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t frameCount,
        callback_t cbf,
        void* user,
        uint32_t notificationFrames,
        bool threadCanCallJava,
        audio_session_t sessionId,
        transfer_type transferType,
        audio_input_flags_t flags,
        uid_t uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        audio_port_handle_t selectedDeviceId,
        audio_microphone_direction_t selectedMicDirection,
        float microphoneFieldDimension)
{
    switch (transferType) {      // 传入的transfer是default，其余case可以先忽略
    case TRANSFER_DEFAULT:
        if (cbf == NULL || threadCanCallJava) {
            transferType = TRANSFER_SYNC;
        } else {
            transferType = TRANSFER_CALLBACK;
        }
        break;
    case TRANSFER_CALLBACK:
        if (cbf == NULL) {
            ALOGE("%s(): Transfer type TRANSFER_CALLBACK but cbf == NULL", __func__);
            status = BAD_VALUE;
            goto exit;
        }
        break;
    case TRANSFER_OBTAIN:
    case TRANSFER_SYNC:
        break;
    default:
        ALOGE("%s(): Invalid transfer type %d", __func__, transferType);
        status = BAD_VALUE;
        goto exit;
    }
    mTransfer = transferType;
    mSampleRate = sampleRate;

    // these below should probably come from the audioFlinger too...
    if (format == AUDIO_FORMAT_DEFAULT) {
        format = AUDIO_FORMAT_PCM_16_BIT;
    }

    // validate parameters
    // AudioFlinger capture only supports linear PCM
    if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) {
        ALOGE("%s(): Format %#x is not linear pcm", __func__, format);
        status = BAD_VALUE;
        goto exit;
    }
    mFormat = format;

    if (!audio_is_input_channel(channelMask)) {
        ALOGE("%s(): Invalid channel mask %#x", __func__, channelMask);
        status = BAD_VALUE;
        goto exit;
    }
    mChannelMask = channelMask;
    channelCount = audio_channel_count_from_in_mask(channelMask);
    mChannelCount = channelCount;

    if (audio_is_linear_pcm(format)) {
        mFrameSize = channelCount * audio_bytes_per_sample(format);
    } else {
        mFrameSize = sizeof(uint8_t);
    }

    // mFrameCount is initialized in createRecord_l
    mReqFrameCount = frameCount;

    mNotificationFramesReq = notificationFrames;
    // mNotificationFramesAct is initialized in createRecord_l

    mSessionId = sessionId;
    ALOGV("%s(): mSessionId %d", __func__, mSessionId);

    callingPid = IPCThreadState::self()->getCallingPid();
    myPid = getpid();
    if (uid == AUDIO_UID_INVALID || (callingPid != myPid)) {
        mClientUid = IPCThreadState::self()->getCallingUid();
    } else {
        mClientUid = uid;
    }
    if (pid == -1 || (callingPid != myPid)) {
        mClientPid = callingPid;
    } else {
        mClientPid = pid;
    }

    mOrigFlags = mFlags = flags;
    mCbf = cbf;

    if (cbf != NULL) {
        mAudioRecordThread = new AudioRecordThread(*this);  // 回调线程，这样采集好数据后，可以主动回调调用方
        mAudioRecordThread->run("AudioRecord", ANDROID_PRIORITY_AUDIO);
        // thread begins in paused state, and will not reference us until start()
    }

    // create the IAudioRecord
    {
        AutoMutex lock(mLock);
        status = createRecord_l(0 /*epoch*/, mOpPackageName); // 创建record
    }
}

// must be called with mLock held
status_t AudioRecord::createRecord_l(const Modulo<uint32_t> &epoch, const String16& opPackageName)
{
    const int64_t beginNs = systemTime();
    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
    IAudioFlinger::CreateRecordInput input;
    IAudioFlinger::CreateRecordOutput output;
    audio_session_t originalSessionId;
    sp<media::IAudioRecord> record;
    void *iMemPointer;
    audio_track_cblk_t* cblk;
    status_t status;
    static const int32_t kMaxCreateAttempts = 3;
    int32_t remainingAttempts = kMaxCreateAttempts;


    // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
    // After fast request is denied, we will request again if IAudioRecord is re-created.

    // Now that we have a reference to an I/O handle and have not yet handed it off to AudioFlinger,
    // we must release it ourselves if anything goes wrong.

    // Client can only express a preference for FAST.  Server will perform additional tests.
    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
        bool useCaseAllowed =
            // any of these use cases:
            // use case 1: callback transfer mode
            (mTransfer == TRANSFER_CALLBACK) ||
            // use case 2: blocking read mode
            // The default buffer capacity at 48 kHz is 2048 frames, or ~42.6 ms.
            // That's enough for double-buffering with our standard 20 ms rule of thumb for
            // the minimum period of a non-SCHED_FIFO thread.
            // This is needed so that AAudio apps can do a low latency non-blocking read from a
            // callback running with SCHED_FIFO.
            (mTransfer == TRANSFER_SYNC) ||
            // use case 3: obtain/release mode
            (mTransfer == TRANSFER_OBTAIN);
        if (!useCaseAllowed) {
            ALOGD("%s(%d): AUDIO_INPUT_FLAG_FAST denied, incompatible transfer = %s",
                  __func__, mPortId,
                  convertTransferToText(mTransfer));
            mFlags = (audio_input_flags_t) (mFlags & ~(AUDIO_INPUT_FLAG_FAST |
                    AUDIO_INPUT_FLAG_RAW));
        }
    }

    input.attr = mAttributes;
    input.config.sample_rate = mSampleRate;
    input.config.channel_mask = mChannelMask;
    input.config.format = mFormat;
    input.clientInfo.clientUid = mClientUid;
    input.clientInfo.clientPid = mClientPid;
    input.clientInfo.clientTid = -1;
    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
        if (mAudioRecordThread != 0) {
            input.clientInfo.clientTid = mAudioRecordThread->getTid();
        }
    }
    input.opPackageName = opPackageName;
    input.riid = mTracker->getRiid();

    input.flags = mFlags;
    // The notification frame count is the period between callbacks, as suggested by the client
    // but moderated by the server.  For record, the calculations are done entirely on server side.
    input.frameCount = mReqFrameCount;
    input.notificationFrameCount = mNotificationFramesReq;
    input.selectedDeviceId = mSelectedDeviceId;
    input.sessionId = mSessionId;
    originalSessionId = mSessionId;

    do {
        record = audioFlinger->createRecord(input, output, &status); // 创建record
        if (status == NO_ERROR) {
            break;
        }
        if (status != FAILED_TRANSACTION || --remainingAttempts <= 0) {
            ALOGE("%s(%d): AudioFlinger could not create record track, status: %d",
                  __func__, mPortId, status);
            goto exit;
        }
        // FAILED_TRANSACTION happens under very specific conditions causing a state mismatch
        // between audio policy manager and audio flinger during the input stream open sequence
        // and can be recovered by retrying.
        // Leave time for race condition to clear before retrying and randomize delay
        // to reduce the probability of concurrent retries in locked steps.
        usleep((20 + rand() % 30) * 10000);
    } while (1);

    ALOG_ASSERT(record != 0);

    // AudioFlinger now owns the reference to the I/O handle,
    // so we are no longer responsible for releasing it.

    mAwaitBoost = false;
    if (output.flags & AUDIO_INPUT_FLAG_FAST) {
        ALOGI("%s(%d): AUDIO_INPUT_FLAG_FAST successful; frameCount %zu -> %zu",
              __func__, mPortId,
              mReqFrameCount, output.frameCount);
        mAwaitBoost = true;
    }
    mFlags = output.flags;
    mRoutedDeviceId = output.selectedDeviceId;
    mSessionId = output.sessionId;
    mSampleRate = output.sampleRate;

    if (output.cblk == 0) {
        ALOGE("%s(%d): Could not get control block", __func__, mPortId);
        status = NO_INIT;
        goto exit;
    }
    // TODO: Using unsecurePointer() has some associated security pitfalls
    //       (see declaration for details).
    //       Either document why it is safe in this case or address the
    //       issue (e.g. by copying).
    iMemPointer = output.cblk ->unsecurePointer(); // 获取共享内存
    if (iMemPointer == NULL) {
        ALOGE("%s(%d): Could not get control block pointer", __func__, mPortId);
        status = NO_INIT;
        goto exit;
    }
    cblk = static_cast<audio_track_cblk_t*>(iMemPointer);

    // Starting address of buffers in shared memory.
    // The buffers are either immediately after the control block,
    // or in a separate area at discretion of server.
    void *buffers;
    if (output.buffers == 0) {
        buffers = cblk + 1;
    } else {
        // TODO: Using unsecurePointer() has some associated security pitfalls
        //       (see declaration for details).
        //       Either document why it is safe in this case or address the
        //       issue (e.g. by copying).
        buffers = output.buffers->unsecurePointer();
        if (buffers == NULL) {
            ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
            status = NO_INIT;
            goto exit;
        }
    }

    // invariant that mAudioRecord != 0 is true only after set() returns successfully
    if (mAudioRecord != 0) {
        IInterface::asBinder(mAudioRecord)->unlinkToDeath(mDeathNotifier, this);
        mDeathNotifier.clear();
    }
    mAudioRecord = record;
    mCblkMemory = output.cblk;
    mBufferMemory = output.buffers;
    IPCThreadState::self()->flushCommands();

    mCblk = cblk;
    // note that output.frameCount is the (possibly revised) value of mReqFrameCount
    if (output.frameCount < mReqFrameCount || (mReqFrameCount == 0 && output.frameCount == 0)) {
        ALOGW("%s(%d): Requested frameCount %zu but received frameCount %zu",
              __func__, output.portId,
              mReqFrameCount,  output.frameCount);
    }

    // Make sure that application is notified with sufficient margin before overrun.
    // The computation is done on server side.
    if (mNotificationFramesReq > 0 && output.notificationFrameCount != mNotificationFramesReq) {
        ALOGW("%s(%d): Server adjusted notificationFrames from %u to %zu for frameCount %zu",
                __func__, output.portId,
                mNotificationFramesReq, output.notificationFrameCount, output.frameCount);
    }
    mNotificationFramesAct = (uint32_t)output.notificationFrameCount;

    //mInput != input includes the case where mInput == AUDIO_IO_HANDLE_NONE for first creation
    if (mDeviceCallback != 0) {
        if (mInput != AUDIO_IO_HANDLE_NONE) {
            AudioSystem::removeAudioDeviceCallback(this, mInput, mPortId);
        }
        AudioSystem::addAudioDeviceCallback(this, output.inputId, output.portId);
    }

    mPortId = output.portId;
    // We retain a copy of the I/O handle, but don't own the reference
    mInput = output.inputId;
    mRefreshRemaining = true;

    mFrameCount = output.frameCount;
    // If IAudioRecord is re-created, don't let the requested frameCount
    // decrease.  This can confuse clients that cache frameCount().
    if (mFrameCount > mReqFrameCount) {
        mReqFrameCount = mFrameCount;
    }

    // update proxy
    mProxy = new AudioRecordClientProxy(cblk, buffers, mFrameCount, mFrameSize); // 共享内存proxy
    mProxy->setEpoch(epoch);
    mProxy->setMinimum(mNotificationFramesAct);

    mDeathNotifier = new DeathNotifier(this);
    IInterface::asBinder(mAudioRecord)->linkToDeath(mDeathNotifier, this);
  
exit:
    mStatus = status;
    // sp<IAudioTrack> track destructor will cause releaseOutput() to be called by AudioFlinger
    return status;
}

// ----------------------------------------------------------------------------

sp<media::IAudioRecord> AudioFlinger::createRecord(const CreateRecordInput& input,
                                                   CreateRecordOutput& output,
                                                   status_t *status)
{
    sp<RecordThread::RecordTrack> recordTrack;
    sp<RecordHandle> recordHandle;
    sp<Client> client;
    status_t lStatus;
    audio_session_t sessionId = input.sessionId;
    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;

    output.cblk.clear();
    output.buffers.clear();
    output.inputId = AUDIO_IO_HANDLE_NONE;

    bool updatePid = (input.clientInfo.clientPid == -1);
    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
    uid_t clientUid = input.clientInfo.clientUid;
    if (!isAudioServerOrMediaServerUid(callingUid)) {
        ALOGW_IF(clientUid != callingUid,
                "%s uid %d tried to pass itself off as %d",
                __FUNCTION__, callingUid, clientUid);
        clientUid = callingUid;
        updatePid = true;
    }
    pid_t clientPid = input.clientInfo.clientPid;
    const pid_t callingPid = IPCThreadState::self()->getCallingPid();
    if (updatePid) {
        ALOGW_IF(clientPid != -1 && clientPid != callingPid,
                 "%s uid %d pid %d tried to pass itself off as pid %d",
                 __func__, callingUid, callingPid, clientPid);
        clientPid = callingPid;
    }

    // we don't yet support anything other than linear PCM
    if (!audio_is_valid_format(input.config.format) || !audio_is_linear_pcm(input.config.format)) {
        ALOGE("createRecord() invalid format %#x", input.config.format);
        lStatus = BAD_VALUE;
        goto Exit;
    }

    // further channel mask checks are performed by createRecordTrack_l()
    if (!audio_is_input_channel(input.config.channel_mask)) {
        ALOGE("createRecord() invalid channel mask %#x", input.config.channel_mask);
        lStatus = BAD_VALUE;
        goto Exit;
    }

    if (sessionId == AUDIO_SESSION_ALLOCATE) {
        sessionId = (audio_session_t) newAudioUniqueId(AUDIO_UNIQUE_ID_USE_SESSION);
    } else if (audio_unique_id_get_use(sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) {
        lStatus = BAD_VALUE;
        goto Exit;
    }

    output.sessionId = sessionId;
    output.selectedDeviceId = input.selectedDeviceId;
    output.flags = input.flags;

    client = registerPid(clientPid);  //创建共享内存

    // Not a conventional loop, but a retry loop for at most two iterations total.
    // Try first maybe with FAST flag then try again without FAST flag if that fails.
    // Exits loop via break on no error of got exit on error
    // The sp<> references will be dropped when re-entering scope.
    // The lack of indentation is deliberate, to reduce code churn and ease merges.
    for (;;) {
    // release previously opened input if retrying.
    if (output.inputId != AUDIO_IO_HANDLE_NONE) {
        recordTrack.clear();
        AudioSystem::releaseInput(portId);
        output.inputId = AUDIO_IO_HANDLE_NONE;
        output.selectedDeviceId = input.selectedDeviceId;
        portId = AUDIO_PORT_HANDLE_NONE;
    }
    lStatus = AudioSystem::getInputForAttr(&input.attr, &output.inputId, // 调用hal层获取属性信息
                                      input.riid,
                                      sessionId,
                                    // FIXME compare to AudioTrack
                                      clientPid,
                                      clientUid,
                                      input.opPackageName,
                                      &input.config,
                                      output.flags, &output.selectedDeviceId, &portId);
    if (lStatus != NO_ERROR) {
        ALOGE("createRecord() getInputForAttr return error %d", lStatus);
        goto Exit;
    }

    {
        Mutex::Autolock _l(mLock);
        RecordThread *thread = checkRecordThread_l(output.inputId); // 获取采集线程，验证猜想2
        if (thread == NULL) {
            ALOGW("createRecord() checkRecordThread_l failed, input handle %d", output.inputId);
            lStatus = FAILED_TRANSACTION;
            goto Exit;
        }

        ALOGV("createRecord() lSessionId: %d input %d", sessionId, output.inputId);

        output.sampleRate = input.config.sample_rate;
        output.frameCount = input.frameCount;
        output.notificationFrameCount = input.notificationFrameCount;

        recordTrack = thread->createRecordTrack_l(client, input.attr, &output.sampleRate, // 在采集线程中创建Record，验证猜想1
                                                  input.config.format, input.config.channel_mask,
                                                  &output.frameCount, sessionId,
                                                  &output.notificationFrameCount,
                                                  callingPid, clientUid, &output.flags,
                                                  input.clientInfo.clientTid,
                                                  &lStatus, portId,
                                                  input.opPackageName);
        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));

        // lStatus == BAD_TYPE means FAST flag was rejected: request a new input from
        // audio policy manager without FAST constraint
        if (lStatus == BAD_TYPE) {
            continue;
        }

        if (lStatus != NO_ERROR) {
            goto Exit;
        }

        // Check if one effect chain was awaiting for an AudioRecord to be created on this
        // session and move it to this thread.
        sp<EffectChain> chain = getOrphanEffectChain_l(sessionId);
        if (chain != 0) {
            Mutex::Autolock _l(thread->mLock);
            thread->addEffectChain_l(chain);
        }
        break;
    }
    // End of retry loop.
    // The lack of indentation is deliberate, to reduce code churn and ease merges.
    }

    output.cblk = recordTrack->getCblk(); 
    output.buffers = recordTrack->getBuffers(); // 返回匿名共享内存操作符
    output.portId = portId;

    // return handle to client
    recordHandle = new RecordHandle(recordTrack); // 包装成binder， 证明了猜想3

Exit:
    if (lStatus != NO_ERROR) {
        // remove local strong reference to Client before deleting the RecordTrack so that the
        // Client destructor is called by the TrackBase destructor with mClientLock held
        // Don't hold mClientLock when releasing the reference on the track as the
        // destructor will acquire it.
        {
            Mutex::Autolock _cl(mClientLock);
            client.clear();
        }
        recordTrack.clear();
        if (output.inputId != AUDIO_IO_HANDLE_NONE) {
            AudioSystem::releaseInput(portId);
        }
    }

    *status = lStatus;
    return recordHandle;
}

// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
        const sp<AudioFlinger::Client>& client,
        const audio_attributes_t& attr,
        uint32_t *pSampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t *pFrameCount,
        audio_session_t sessionId,
        size_t *pNotificationFrameCount,
        pid_t creatorPid,
        uid_t uid,
        audio_input_flags_t *flags,
        pid_t tid,
        status_t *status,
        audio_port_handle_t portId,
        const String16& opPackageName)
{
    size_t frameCount = *pFrameCount;
    size_t notificationFrameCount = *pNotificationFrameCount;
    sp<RecordTrack> track;
    status_t lStatus;
    audio_input_flags_t inputFlags = mInput->flags;
    audio_input_flags_t requestedFlags = *flags;
    uint32_t sampleRate;

    lStatus = initCheck();

    if (*pSampleRate == 0) {
        *pSampleRate = mSampleRate;
    }
    sampleRate = *pSampleRate;

    // special case for FAST flag considered OK if fast capture is present
    if (hasFastCapture()) {
        inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST);
    }

    // Check if requested flags are compatible with input stream flags
    if ((*flags & inputFlags) != *flags) {
        ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and"
                " input flags (%08x)",
              *flags, inputFlags);
        *flags = (audio_input_flags_t)(*flags & inputFlags);
    }

    // client expresses a preference for FAST, but we get the final say
    if (*flags & AUDIO_INPUT_FLAG_FAST) {
      if (
            // we formerly checked for a callback handler (non-0 tid),
            // but that is no longer required for TRANSFER_OBTAIN mode
            //
            // Frame count is not specified (0), or is less than or equal the pipe depth.
            // It is OK to provide a higher capacity than requested.
            // We will force it to mPipeFramesP2 below.
            (frameCount <= mPipeFramesP2) &&
            // PCM data
            audio_is_linear_pcm(format) &&
            // hardware format
            (format == mFormat) &&
            // hardware channel mask
            (channelMask == mChannelMask) &&
            // hardware sample rate
            (sampleRate == mSampleRate) &&
            // record thread has an associated fast capture
            hasFastCapture() &&
            // there are sufficient fast track slots available
            mFastTrackAvail
        ) {
          // check compatibility with audio effects.
          Mutex::Autolock _l(mLock);
          // Do not accept FAST flag if the session has software effects
          sp<EffectChain> chain = getEffectChain_l(sessionId);
          if (chain != 0) {
              audio_input_flags_t old = *flags;
              chain->checkInputFlagCompatibility(flags);
              if (old != *flags) {
                  ALOGV("%p AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x",
                          this, (int)old, (int)*flags);
              }
          }
          ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0,
                   "%p AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu",
                   this, frameCount, mFrameCount);
      } else {
        ALOGV("%p AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu "
                "format=%#x isLinear=%d mFormat=%#x channelMask=%#x sampleRate=%u mSampleRate=%u "
                "hasFastCapture=%d tid=%d mFastTrackAvail=%d",
                this, frameCount, mFrameCount, mPipeFramesP2,
                format, audio_is_linear_pcm(format), mFormat, channelMask, sampleRate, mSampleRate,
                hasFastCapture(), tid, mFastTrackAvail);
        *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST);
      }
    }

    // If FAST or RAW flags were corrected, ask caller to request new input from audio policy
    if ((*flags & AUDIO_INPUT_FLAG_FAST) !=
            (requestedFlags & AUDIO_INPUT_FLAG_FAST)) {
        *flags = (audio_input_flags_t) (*flags & ~(AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW));
        lStatus = BAD_TYPE;
        goto Exit;
    }

    // compute track buffer size in frames, and suggest the notification frame count
    if (*flags & AUDIO_INPUT_FLAG_FAST) {
        // fast track: frame count is exactly the pipe depth
        frameCount = mPipeFramesP2;
        // ignore requested notificationFrames, and always notify exactly once every HAL buffer
        notificationFrameCount = mFrameCount;
    } else {
        // not fast track: max notification period is resampled equivalent of one HAL buffer time
        //                 or 20 ms if there is a fast capture
        // TODO This could be a roundupRatio inline, and const
        size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount)
                * sampleRate + mSampleRate - 1) / mSampleRate;
        // minimum number of notification periods is at least kMinNotifications,
        // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs)
        static const size_t kMinNotifications = 3;
        static const uint32_t kMinMs = 30;
        // TODO This could be a roundupRatio inline
        const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000;
        // TODO This could be a roundupRatio inline
        const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) /
                maxNotificationFrames;
        const size_t minFrameCount = maxNotificationFrames *
                max(kMinNotifications, minNotificationsByMs);
        frameCount = max(frameCount, minFrameCount);
        if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) {
            notificationFrameCount = maxNotificationFrames;
        }
    }
    *pFrameCount = frameCount;
    *pNotificationFrameCount = notificationFrameCount;

    { // scope for mLock
        Mutex::Autolock _l(mLock);

        track = new RecordTrack(this, client, attr, sampleRate, // 创建Record
                      format, channelMask, frameCount,
                      nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid, uid,
                      *flags, TrackBase::TYPE_DEFAULT, opPackageName, portId);

        lStatus = track->initCheck();
        if (lStatus != NO_ERROR) {
            ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus);
            // track must be cleared from the caller as the caller has the AF lock
            goto Exit;
        }
        mTracks.add(track); // 将Record加入到列表中

        if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) {
            pid_t callingPid = IPCThreadState::self()->getCallingPid();
            // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful,
            // so ask activity manager to do this on our behalf
            sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/);
        }
    }

    lStatus = NO_ERROR;

Exit:
    *status = lStatus;
    return track;
}

// RecordTrack constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
            RecordThread *thread,
            const sp<Client>& client,
            const audio_attributes_t& attr,
            uint32_t sampleRate,
            audio_format_t format,
            audio_channel_mask_t channelMask,
            size_t frameCount,
            void *buffer,
            size_t bufferSize,
            audio_session_t sessionId,
            pid_t creatorPid,
            uid_t uid,
            audio_input_flags_t flags,
            track_type type,
            const String16& opPackageName,
            audio_port_handle_t portId)
    :   TrackBase(thread, client, attr, sampleRate, format,
                  channelMask, frameCount, buffer, bufferSize, sessionId,
                  creatorPid, uid, false /*isOut*/,
                  (type == TYPE_DEFAULT) ?
                          ((flags & AUDIO_INPUT_FLAG_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
                          ((buffer == NULL) ? ALLOC_LOCAL : ALLOC_NONE),
                  type, portId,
                  std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_RECORD) + std::to_string(portId)),
        mOverflow(false),
        mFramesToDrop(0),
        mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
        mRecordBufferConverter(NULL),
        mFlags(flags),
        mSilenced(false),
        mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(uid, attr, opPackageName))
{
    if (mCblk == NULL) {
        return;
    }

    if (!isDirect()) {
        mRecordBufferConverter = new RecordBufferConverter(
                thread->mChannelMask, thread->mFormat, thread->mSampleRate,
                channelMask, format, sampleRate);
        // Check if the RecordBufferConverter construction was successful.
        // If not, don't continue with construction.
        //
        // NOTE: It would be extremely rare that the record track cannot be created
        // for the current device, but a pending or future device change would make
        // the record track configuration valid.
        if (mRecordBufferConverter->initCheck() != NO_ERROR) {
            ALOGE("%s(%d): RecordTrack unable to create record buffer converter", __func__, mId);
            return;
        }
    }

    mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
            mFrameSize, !isExternalTrack());

    mResamplerBufferProvider = new ResamplerBufferProvider(this);

    if (flags & AUDIO_INPUT_FLAG_FAST) {
        ALOG_ASSERT(thread->mFastTrackAvail);
        thread->mFastTrackAvail = false;
    } else {
        // TODO: only Normal Record has timestamps (Fast Record does not).
        mServerLatencySupported = checkServerLatencySupported(mFormat, flags);
    }
#ifdef TEE_SINK
    mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
            + "_" + std::to_string(mId)
            + "_R");
#endif

    // Once this item is logged by the server, the client can add properties.
    mTrackMetrics.logConstructor(creatorPid, uid);
}

AudioFlinger::ThreadBase::TrackBase::TrackBase(
            ThreadBase *thread,
            const sp<Client>& client,
            const audio_attributes_t& attr,
            uint32_t sampleRate,
            audio_format_t format,
            audio_channel_mask_t channelMask,
            size_t frameCount,
            void *buffer,
            size_t bufferSize,
            audio_session_t sessionId,
            pid_t creatorPid,
            uid_t clientUid,
            bool isOut,
            alloc_type alloc,
            track_type type,
            audio_port_handle_t portId,
            std::string metricsId)
    :   RefBase(),
        mThread(thread),
        mClient(client),
        mCblk(NULL),
        // mBuffer, mBufferSize
        mState(IDLE),
        mAttr(attr),
        mSampleRate(sampleRate),
        mFormat(format),
        mChannelMask(channelMask),
        mChannelCount(isOut ?
                audio_channel_count_from_out_mask(channelMask) :
                audio_channel_count_from_in_mask(channelMask)),
        mFrameSize(audio_has_proportional_frames(format) ?
                mChannelCount * audio_bytes_per_sample(format) : sizeof(int8_t)),
        mFrameCount(frameCount),
        mSessionId(sessionId),
        mIsOut(isOut),
        mId(android_atomic_inc(&nextTrackId)),
        mTerminated(false),
        mType(type),
        mThreadIoHandle(thread ? thread->id() : AUDIO_IO_HANDLE_NONE),
        mPortId(portId),
        mIsInvalid(false),
        mTrackMetrics(std::move(metricsId), isOut),
        mCreatorPid(creatorPid)
{
    const uid_t callingUid = IPCThreadState::self()->getCallingUid();
    mUid = clientUid;
    size_t minBufferSize = buffer == NULL ? roundup(frameCount) : frameCount;
    // check overflow when computing bufferSize due to multiplication by mFrameSize.
    if (minBufferSize < frameCount  // roundup rounds down for values above UINT_MAX / 2
            || mFrameSize == 0   // format needs to be correct
            || minBufferSize > SIZE_MAX / mFrameSize) {
        android_errorWriteLog(0x534e4554, "34749571");
        return;
    }
    minBufferSize *= mFrameSize;

    if (buffer == nullptr) {
        bufferSize = minBufferSize; // allocated here.
    } else if (minBufferSize > bufferSize) {
        android_errorWriteLog(0x534e4554, "38340117");
        return;
    }

    size_t size = sizeof(audio_track_cblk_t);
    if (buffer == NULL && alloc == ALLOC_CBLK) {
        // check overflow when computing allocation size for streaming tracks.
        if (size > SIZE_MAX - bufferSize) {
            android_errorWriteLog(0x534e4554, "34749571");
            return;
        }
        size += bufferSize;
    }

    if (client != 0) {
        mCblkMemory = client->heap()->allocate(size);  // 分配共享内存，client携带了创建的共享内存信息
        if (mCblkMemory == 0 ||
                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->unsecurePointer())) == NULL) {
            ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
            client->heap()->dump("AudioTrack");
            mCblkMemory.clear();
            return;
        }
    } else {
        mCblk = (audio_track_cblk_t *) malloc(size);
        if (mCblk == NULL) {
            ALOGE("%s(%d): not enough memory for AudioTrack size=%zu", __func__, mId, size);
            return;
        }
    }

    // construct the shared structure in-place.
    if (mCblk != NULL) {
        new(mCblk) audio_track_cblk_t();
        switch (alloc) {
        case ALLOC_READONLY: {
            const sp<MemoryDealer> roHeap(thread->readOnlyHeap());
            if (roHeap == 0 ||
                    (mBufferMemory = roHeap->allocate(bufferSize)) == 0 ||
                    (mBuffer = mBufferMemory->unsecurePointer()) == NULL) {
                ALOGE("%s(%d): not enough memory for read-only buffer size=%zu",
                        __func__, mId, bufferSize);
                if (roHeap != 0) {
                    roHeap->dump("buffer");
                }
                mCblkMemory.clear();
                mBufferMemory.clear();
                return;
            }
            memset(mBuffer, 0, bufferSize);
            } break;
        case ALLOC_PIPE:
            mBufferMemory = thread->pipeMemory();
            // mBuffer is the virtual address as seen from current process (mediaserver),
            // and should normally be coming from mBufferMemory->unsecurePointer().
            // However in this case the TrackBase does not reference the buffer directly.
            // It should references the buffer via the pipe.
            // Therefore, to detect incorrect usage of the buffer, we set mBuffer to NULL.
            mBuffer = NULL;
            bufferSize = 0;
            break;
        case ALLOC_CBLK:
            // clear all buffers
            if (buffer == NULL) {
                mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
                memset(mBuffer, 0, bufferSize);
            } else {
                mBuffer = buffer;
#if 0
                mCblk->mFlags = CBLK_FORCEREADY;    // FIXME hack, need to fix the track ready logic
#endif
            }
            break;
        case ALLOC_LOCAL:
            mBuffer = calloc(1, bufferSize);
            break;
        case ALLOC_NONE:
            mBuffer = buffer;
            break;
        default:
            LOG_ALWAYS_FATAL("%s(%d): invalid allocation type: %d", __func__, mId, (int)alloc);
        }
        mBufferSize = bufferSize;

#ifdef TEE_SINK
        mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
#endif

    }
}

sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
{
    Mutex::Autolock _cl(mClientLock);
    // If pid is already in the mClients wp<> map, then use that entry
    // (for which promote() is always != 0), otherwise create a new entry and Client.
    sp<Client> client = mClients.valueFor(pid).promote();
    if (client == 0) {
        client = new Client(this, pid);
        mClients.add(pid, client);
    }

    return client;
}

AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
    :   RefBase(),
        mAudioFlinger(audioFlinger),
        mPid(pid)
{
    mMemoryDealer = new MemoryDealer(
            audioFlinger->getClientSharedHeapSize(),
            (std::string("AudioFlinger::Client(") + std::to_string(pid) + ")").c_str());
}

AudioFlinger::RecordHandle::~RecordHandle() {
    stop_nonvirtual();
    mRecordTrack->destroy();
}

binder::Status AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
        int /*audio_session_t*/ triggerSession) {
    ALOGV("%s()", __func__);
    return binder::Status::fromStatusT(
        mRecordTrack->start((AudioSystem::sync_event_t)event, (audio_session_t) triggerSession));
}

binder::Status AudioFlinger::RecordHandle::stop() {
    stop_nonvirtual();
    return binder::Status::ok();
}

void AudioFlinger::RecordHandle::stop_nonvirtual() {
    ALOGV("%s()", __func__);
    mRecordTrack->stop();
}

binder::Status AudioFlinger::RecordHandle::getActiveMicrophones(
        std::vector<media::MicrophoneInfo>* activeMicrophones) {
    ALOGV("%s()", __func__);
    return binder::Status::fromStatusT(
            mRecordTrack->getActiveMicrophones(activeMicrophones));
}

binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneDirection(
        int /*audio_microphone_direction_t*/ direction) {
    ALOGV("%s()", __func__);
    return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneDirection(
            static_cast<audio_microphone_direction_t>(direction)));
}

binder::Status AudioFlinger::RecordHandle::setPreferredMicrophoneFieldDimension(float zoom) {
    ALOGV("%s()", __func__);
    return binder::Status::fromStatusT(mRecordTrack->setPreferredMicrophoneFieldDimension(zoom));
}