AudioTrack的源码解读(1)

 public AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
            int mode, int sessionId)
                    throws IllegalArgumentException {
        this(attributes, format, bufferSizeInBytes, mode, sessionId, false /*offload*/,
                ENCAPSULATION_MODE_NONE, null /* tunerConfiguration */);
    }

    public final static int[] SDK_USAGES = {
            USAGE_UNKNOWN,
            USAGE_MEDIA,
            USAGE_VOICE_COMMUNICATION,
            USAGE_VOICE_COMMUNICATION_SIGNALLING,
            USAGE_ALARM,
            USAGE_NOTIFICATION,
            USAGE_NOTIFICATION_RINGTONE,
            USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
            USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
            USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
            USAGE_NOTIFICATION_EVENT,
            USAGE_ASSISTANCE_ACCESSIBILITY,
            USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
            USAGE_ASSISTANCE_SONIFICATION,
            USAGE_GAME,
            USAGE_ASSISTANT,
    };

        public Builder setContentType(@AttributeContentType int contentType) {
            switch (contentType) {
                case CONTENT_TYPE_UNKNOWN:
                case CONTENT_TYPE_MOVIE:
                case CONTENT_TYPE_MUSIC:
                case CONTENT_TYPE_SONIFICATION:
                case CONTENT_TYPE_SPEECH:
                    mContentType = contentType;
                    break;
                default:
                    throw new IllegalArgumentException("Invalid content type " + contentType);
            }
            return this;
        }

 FLAG_ALL = FLAG_AUDIBILITY_ENFORCED | FLAG_SECURE | FLAG_SCO
            | FLAG_BEACON | FLAG_HW_AV_SYNC | FLAG_HW_HOTWORD | FLAG_BYPASS_INTERRUPTION_POLICY
            | FLAG_BYPASS_MUTE | FLAG_LOW_LATENCY | FLAG_DEEP_BUFFER | FLAG_NO_MEDIA_PROJECTION
            | FLAG_NO_SYSTEM_CAPTURE | FLAG_CAPTURE_PRIVATE;

    static public int getMinBufferSize(int sampleRateInHz, int channelConfig, int audioFormat) {
        int channelCount = 0;
        switch(channelConfig) {     // 转换声道数量
        case AudioFormat.CHANNEL_OUT_MONO:
        case AudioFormat.CHANNEL_CONFIGURATION_MONO:
            channelCount = 1;
            break;
        case AudioFormat.CHANNEL_OUT_STEREO:
        case AudioFormat.CHANNEL_CONFIGURATION_STEREO:
            channelCount = 2;
            break;
        default:
            if (!isMultichannelConfigSupported(channelConfig, audioFormat)) {
                loge("getMinBufferSize(): Invalid channel configuration.");
                return ERROR_BAD_VALUE;
            } else {
                channelCount = AudioFormat.channelCountFromOutChannelMask(channelConfig);
            }
        }
 // 格式检查，可以看到除了PCM格式，还有MP3，OPUS等
        if (!AudioFormat.isPublicEncoding(audioFormat)) {
            loge("getMinBufferSize(): Invalid audio format.");
            return ERROR_BAD_VALUE;
        }

       // 频率合法性检查，通过JNI下沉到Native
        // sample rate, note these values are subject to change
        // Note: AudioFormat.SAMPLE_RATE_UNSPECIFIED is not allowed 
        if ( (sampleRateInHz < AudioFormat.SAMPLE_RATE_HZ_MIN) ||
                (sampleRateInHz > AudioFormat.SAMPLE_RATE_HZ_MAX) ) {
            loge("getMinBufferSize(): " + sampleRateInHz + " Hz is not a supported sample rate.");
            return ERROR_BAD_VALUE;
        }

        int size = native_get_min_buff_size(sampleRateInHz, channelCount, audioFormat);
        if (size <= 0) {
            loge("getMinBufferSize(): error querying hardware");
            return ERROR;
        }
        else {
            return size;
        }
    }

static jint android_media_AudioTrack_get_min_buff_size(JNIEnv *env,  jobject thiz,
    jint sampleRateInHertz, jint channelCount, jint audioFormat) {

    size_t frameCount;
    const status_t status = AudioTrack::getMinFrameCount(&frameCount, AUDIO_STREAM_DEFAULT,
            sampleRateInHertz); // 获取最小帧数
    if (status != NO_ERROR) {
        ALOGE("AudioTrack::getMinFrameCount() for sample rate %d failed with status %d",
                sampleRateInHertz, status);
        return -1;
    }
    const audio_format_t format = audioFormatToNative(audioFormat);
    if (audio_has_proportional_frames(format)) {
        const size_t bytesPerSample = audio_bytes_per_sample(format);
        return frameCount * channelCount * bytesPerSample;
    } else {
        return frameCount;
    }
}

 private AudioTrack(AudioAttributes attributes, AudioFormat format, int bufferSizeInBytes,
            int mode, int sessionId, boolean offload, int encapsulationMode,
            @Nullable TunerConfiguration tunerConfiguration)
                    throws IllegalArgumentException {
        super(attributes, AudioPlaybackConfiguration.PLAYER_TYPE_JAM_AUDIOTRACK);
        // mState already == STATE_UNINITIALIZED

        mConfiguredAudioAttributes = attributes; // object copy not needed, immutable.
        // Check if we should enable deep buffer mode
        if (shouldEnablePowerSaving(mAttributes, format, bufferSizeInBytes, mode)) {
            mAttributes = new AudioAttributes.Builder(mAttributes)
                .replaceFlags((mAttributes.getAllFlags()
                        | AudioAttributes.FLAG_DEEP_BUFFER)
                        & ~AudioAttributes.FLAG_LOW_LATENCY)
                .build();
        }
        int rate = format.getSampleRate();
        if (rate == AudioFormat.SAMPLE_RATE_UNSPECIFIED) {
            rate = 0;
        }

        int channelIndexMask = 0;
        if ((format.getPropertySetMask()
                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_INDEX_MASK) != 0) {
            channelIndexMask = format.getChannelIndexMask();
        }
        int channelMask = 0;
        if ((format.getPropertySetMask()
                & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_CHANNEL_MASK) != 0) {
            channelMask = format.getChannelMask();
        } else if (channelIndexMask == 0) { // if no masks at all, use stereo
            channelMask = AudioFormat.CHANNEL_OUT_FRONT_LEFT
                    | AudioFormat.CHANNEL_OUT_FRONT_RIGHT;
        }
        int encoding = AudioFormat.ENCODING_DEFAULT;
        if ((format.getPropertySetMask() & AudioFormat.AUDIO_FORMAT_HAS_PROPERTY_ENCODING) != 0) {
            encoding = format.getEncoding();
        }
        audioParamCheck(rate, channelMask, channelIndexMask, encoding, mode);
        mOffloaded = offload;
        mStreamType = AudioSystem.STREAM_DEFAULT;

        audioBuffSizeCheck(bufferSizeInBytes);

        mInitializationLooper = looper;

        if (sessionId < 0) {
            throw new IllegalArgumentException("Invalid audio session ID: "+sessionId);
        }

        int[] sampleRate = new int[] {mSampleRate};
        int[] session = new int[1];
        session[0] = sessionId;
        // native initialization
        int initResult = native_setup(new WeakReference<AudioTrack>(this), mAttributes,
                sampleRate, mChannelMask, mChannelIndexMask, mAudioFormat,
                mNativeBufferSizeInBytes, mDataLoadMode, session, 0 /*nativeTrackInJavaObj*/,
                offload, encapsulationMode, tunerConfiguration,
                getCurrentOpPackageName());
        if (initResult != SUCCESS) {
            loge("Error code "+initResult+" when initializing AudioTrack.");
            return; // with mState == STATE_UNINITIALIZED
        }

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

        // TODO: consider caching encapsulationMode and tunerConfiguration in the Java object.

        if ((mAttributes.getFlags() & AudioAttributes.FLAG_HW_AV_SYNC) != 0) {
            int frameSizeInBytes;
            if (AudioFormat.isEncodingLinearFrames(mAudioFormat)) {
                frameSizeInBytes = mChannelCount * AudioFormat.getBytesPerSample(mAudioFormat);
            } else {
                frameSizeInBytes = 1;
            }
            mOffset = ((int) Math.ceil(HEADER_V2_SIZE_BYTES / frameSizeInBytes)) * frameSizeInBytes;
        }

        if (mDataLoadMode == MODE_STATIC) {
            mState = STATE_NO_STATIC_DATA;
        } else {
            mState = STATE_INITIALIZED;
        }

        baseRegisterPlayer();
    }

static jint android_media_AudioTrack_setup(JNIEnv *env, jobject thiz, jobject weak_this,
                                           jobject jaa, jintArray jSampleRate,
                                           jint channelPositionMask, jint channelIndexMask,
                                           jint audioFormat, jint buffSizeInBytes, jint memoryMode,
                                           jintArray jSession, jlong nativeAudioTrack,
                                           jboolean offload, jint encapsulationMode,
                                           jobject tunerConfiguration, jstring opPackageName)
                                           {
// 如果Native对应的AudioTrack还没创建，那么就需要先创建Native的AudioTrack
  if (nativeAudioTrack == 0) {
    // create the native AudioTrack object
        ScopedUtfChars opPackageNameStr(env, opPackageName);
        lpTrack = new AudioTrack(opPackageNameStr.c_str());

       // 创建回调结构，用来传递数据
        // initialize the callback information:
        // this data will be passed with every AudioTrack callback
        lpJniStorage = new AudioTrackJniStorage();
        lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
        // we use a weak reference so the AudioTrack object can be garbage collected.
        lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
        lpJniStorage->mCallbackData.isOffload = offload;
        lpJniStorage->mCallbackData.busy = false;

 // 分别处理Static和Stream模式，可以看到对音频数据的内存处理有明显差异。
       // initialize the native AudioTrack object
        status_t status = NO_ERROR;
        switch (memoryMode) {
        case MODE_STREAM:
            status = lpTrack->set(AUDIO_STREAM_DEFAULT, // stream type, but more info conveyed
                                                        // in paa (last argument)
                                  sampleRateInHertz,
                                  format, // word length, PCM
                                  nativeChannelMask, offload ? 0 : frameCount,
                                  offload ? AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
                                          : AUDIO_OUTPUT_FLAG_NONE,
                                  audioCallback,
                                  &(lpJniStorage->mCallbackData), // callback, callback data (user)
                                  0,    // notificationFrames == 0 since not using EVENT_MORE_DATA
                                        // to feed the AudioTrack
                                  0,    // shared mem
                                  true, // thread can call Java
                                  sessionId, // audio session ID
                                  offload ? AudioTrack::TRANSFER_SYNC_NOTIF_CALLBACK
                                          : AudioTrack::TRANSFER_SYNC,
                                  (offload || encapsulationMode) ? &offloadInfo : NULL, -1,
                                  -1, // default uid, pid values
                                  paa.get());
            break;

        case MODE_STATIC:
            // AudioTrack is using shared memory

            if (!lpJniStorage->allocSharedMem(buffSizeInBytes)) {
                ALOGE("Error creating AudioTrack in static mode: error creating mem heap base");
                goto native_init_failure;
            }

            status = lpTrack->set(AUDIO_STREAM_DEFAULT, // stream type, but more info conveyed
                                                        // in paa (last argument)
                                  sampleRateInHertz,
                                  format, // word length, PCM
                                  nativeChannelMask, frameCount, AUDIO_OUTPUT_FLAG_NONE,
                                  audioCallback,
                                  &(lpJniStorage->mCallbackData), // callback, callback data (user)
                                  0, // notificationFrames == 0 since not using EVENT_MORE_DATA
                                     // to feed the AudioTrack
                                  lpJniStorage->mMemBase, // shared mem
                                  true,                   // thread can call Java
                                  sessionId,              // audio session ID
                                  AudioTrack::TRANSFER_SHARED,
                                  NULL,   // default offloadInfo
                                  -1, -1, // default uid, pid values
                                  paa.get());
            break;

        default:
            ALOGE("Unknown mode %d", memoryMode);
            goto native_init_failure;
        }
} 

// 如果Native有对应的AudioTrack，需要做的事情就是将Native的AudioTrack对象关联到Java的AudioTrack
  // end if (nativeAudioTrack == 0)
        lpTrack = (AudioTrack*)nativeAudioTrack;
        lpJniStorage = new AudioTrackJniStorage();
        lpJniStorage->mCallbackData.audioTrack_class = (jclass)env->NewGlobalRef(clazz);
        // we use a weak reference so the AudioTrack object can be garbage collected.
        lpJniStorage->mCallbackData.audioTrack_ref = env->NewGlobalRef(weak_this);
        lpJniStorage->mCallbackData.busy = false;
    }
    lpJniStorage->mAudioTrackCallback =
            new JNIAudioTrackCallback(env, thiz, lpJniStorage->mCallbackData.audioTrack_ref,
                                      javaAudioTrackFields.postNativeEventInJava);
    lpTrack->setAudioTrackCallback(lpJniStorage->mAudioTrackCallback);

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

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

    {   // scope for the lock
        Mutex::Autolock l(sLock);
        sAudioTrackCallBackCookies.add(&lpJniStorage->mCallbackData);
    }
    // save our newly created C++ AudioTrack in the "nativeTrackInJavaObj" field
    // of the Java object (in mNativeTrackInJavaObj)
    setAudioTrack(env, thiz, lpTrack);

    // save the JNI resources so we can free them later
    //ALOGV("storing lpJniStorage: %x\n", (long)lpJniStorage);
    env->SetLongField(thiz, javaAudioTrackFields.jniData, (jlong)lpJniStorage);

    // since we had audio attributes, the stream type was derived from them during the
    // creation of the native AudioTrack: push the same value to the Java object
    env->SetIntField(thiz, javaAudioTrackFields.fieldStreamType, (jint) lpTrack->streamType());

    return (jint) AUDIO_JAVA_SUCCESS;
                                           }

status_t AudioTrack::set(
        audio_stream_type_t streamType,
        uint32_t sampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t frameCount,
        audio_output_flags_t flags,
        callback_t cbf,
        void* user,
        int32_t notificationFrames,
        const sp<IMemory>& sharedBuffer,
        bool threadCanCallJava,
        audio_session_t sessionId,
        transfer_type transferType,
        const audio_offload_info_t *offloadInfo,
        uid_t uid,
        pid_t pid,
        const audio_attributes_t* pAttributes,
        bool doNotReconnect,
        float maxRequiredSpeed,
        audio_port_handle_t selectedDeviceId)
{
...

    // handle default values first. 默认streamType
    if (streamType == AUDIO_STREAM_DEFAULT) {
        streamType = AUDIO_STREAM_MUSIC;
    }
    if (pAttributes == NULL) {
        if (uint32_t(streamType) >= AUDIO_STREAM_PUBLIC_CNT) {
            ALOGE("%s(): Invalid stream type %d", __func__, streamType);
            status = BAD_VALUE;
            goto exit;
        }
        mStreamType = streamType;

    } else {
        // stream type shouldn't be looked at, this track has audio attributes
        memcpy(&mAttributes, pAttributes, sizeof(audio_attributes_t));
        ALOGV("%s(): Building AudioTrack with attributes:"
                " usage=%d content=%d flags=0x%x tags=[%s]",
                __func__,
                 mAttributes.usage, mAttributes.content_type, mAttributes.flags, mAttributes.tags);
        mStreamType = AUDIO_STREAM_DEFAULT;
        audio_flags_to_audio_output_flags(mAttributes.flags, &flags);
    }

    // these below should probably come from the audioFlinger too...
    if (format == AUDIO_FORMAT_DEFAULT) {
        format = AUDIO_FORMAT_PCM_16_BIT;
    } else if (format == AUDIO_FORMAT_IEC61937) { // HDMI pass-through?
        flags = static_cast<audio_output_flags_t>(flags | AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO);
    }

    // validate parameters 
//格式检查
    if (!audio_is_valid_format(format)) {
        ALOGE("%s(): Invalid format %#x", __func__, format);
        status = BAD_VALUE;
        goto exit;
    }
    mFormat = format;
// 通道检查
    if (!audio_is_output_channel(channelMask)) {
        ALOGE("%s(): Invalid channel mask %#x",  __func__, channelMask);
        status = BAD_VALUE;
        goto exit;
    }
    mChannelMask = channelMask;
    channelCount = audio_channel_count_from_out_mask(channelMask);
    mChannelCount = channelCount;

    // force direct flag if format is not linear PCM
    // or offload was requested
    if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
            || !audio_is_linear_pcm(format)) {
        ALOGV( (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
                    ? "%s(): Offload request, forcing to Direct Output"
                    : "%s(): Not linear PCM, forcing to Direct Output",
                    __func__);
        flags = (audio_output_flags_t)
                // FIXME why can't we allow direct AND fast?
                ((flags | AUDIO_OUTPUT_FLAG_DIRECT) & ~AUDIO_OUTPUT_FLAG_FAST);
    }

    // force direct flag if HW A/V sync requested
    if ((flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
        flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_DIRECT);
    }

    if (flags & AUDIO_OUTPUT_FLAG_DIRECT) {
        if (audio_has_proportional_frames(format)) {
            mFrameSize = channelCount * audio_bytes_per_sample(format);
        } else {
            mFrameSize = sizeof(uint8_t);
        }
    } else {
        ALOG_ASSERT(audio_has_proportional_frames(format));
        mFrameSize = channelCount * audio_bytes_per_sample(format);
        // createTrack will return an error if PCM format is not supported by server,
        // so no need to check for specific PCM formats here
    }

    // sampling rate must be specified for direct outputs
    if (sampleRate == 0 && (flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
        status = BAD_VALUE;
        goto exit;
    }
    mSampleRate = sampleRate;
    mOriginalSampleRate = sampleRate;
    mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
    // 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
    mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);

    // Make copy of input parameter offloadInfo so that in the future:
    //  (a) createTrack_l doesn't need it as an input parameter
    //  (b) we can support re-creation of offloaded tracks
    if (offloadInfo != NULL) {
        mOffloadInfoCopy = *offloadInfo;
        mOffloadInfo = &mOffloadInfoCopy;
    } else {
        mOffloadInfo = NULL;
        memset(&mOffloadInfoCopy, 0, sizeof(audio_offload_info_t));
    }

    mVolume[AUDIO_INTERLEAVE_LEFT] = 1.0f;
    mVolume[AUDIO_INTERLEAVE_RIGHT] = 1.0f;
    mSendLevel = 0.0f;
    // mFrameCount is initialized in createTrack_l
    mReqFrameCount = frameCount;
    if (notificationFrames >= 0) {
        mNotificationFramesReq = notificationFrames;
        mNotificationsPerBufferReq = 0;
    } else {
        if (!(flags & AUDIO_OUTPUT_FLAG_FAST)) {
            ALOGE("%s(): notificationFrames=%d not permitted for non-fast track",
                    __func__, notificationFrames);
            status = BAD_VALUE;
            goto exit;
        }
        if (frameCount > 0) {
            ALOGE("%s(): notificationFrames=%d not permitted with non-zero frameCount=%zu",
                    __func__, notificationFrames, frameCount);
            status = BAD_VALUE;
            goto exit;
        }
        mNotificationFramesReq = 0;
        const uint32_t minNotificationsPerBuffer = 1;
        const uint32_t maxNotificationsPerBuffer = 8;
        mNotificationsPerBufferReq = min(maxNotificationsPerBuffer,
                max((uint32_t) -notificationFrames, minNotificationsPerBuffer));
        ALOGW_IF(mNotificationsPerBufferReq != (uint32_t) -notificationFrames,
                "%s(): notificationFrames=%d clamped to the range -%u to -%u",
                __func__,
                notificationFrames, minNotificationsPerBuffer, maxNotificationsPerBuffer);
    }
    mNotificationFramesAct = 0;
    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;
    }
    mAuxEffectId = 0;
    mOrigFlags = mFlags = flags;
    mCbf = cbf;

    if (cbf != NULL) {
        mAudioTrackThread = new AudioTrackThread(*this);
        mAudioTrackThread->run("AudioTrack", ANDROID_PRIORITY_AUDIO, 0 /*stack*/);
        // thread begins in paused state, and will not reference us until start()
    }

    // create the IAudioTrack
    {
        AutoMutex lock(mLock);
        status = createTrack_l();
    }
 ...
// 接下来是一些参数设置，可以忽略
}

status_t AudioTrack::createTrack_l()
{
    status_t status;
    bool callbackAdded = false;

// 这儿就出现了AudioFlinger
    const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
    {
// 接下来就是打包参数给flinger
    // mFlags (not mOrigFlags) is modified depending on whether fast request is accepted.
    // After fast request is denied, we will request again if IAudioTrack is re-created.
    // Client can only express a preference for FAST.  Server will perform additional tests.
    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
        // either of these use cases:
        // use case 1: shared buffer
        bool sharedBuffer = mSharedBuffer != 0;
        bool transferAllowed =
            // use case 2: callback transfer mode
            (mTransfer == TRANSFER_CALLBACK) ||
            // use case 3: obtain/release mode
            (mTransfer == TRANSFER_OBTAIN) ||
            // use case 4: synchronous write
            ((mTransfer == TRANSFER_SYNC || mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK)
                    && mThreadCanCallJava);

        bool fastAllowed = sharedBuffer || transferAllowed;
        if (!fastAllowed) {
            ALOGW("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by client,"
                  " not shared buffer and transfer = %s",
                  __func__, mPortId,
                  convertTransferToText(mTransfer));
            mFlags = (audio_output_flags_t) (mFlags & ~AUDIO_OUTPUT_FLAG_FAST);
        }
    }

    IAudioFlinger::CreateTrackInput input;
    if (mStreamType != AUDIO_STREAM_DEFAULT) {
        input.attr = AudioSystem::streamTypeToAttributes(mStreamType);
    } else {
        input.attr = mAttributes;
    }
    input.config = AUDIO_CONFIG_INITIALIZER;
    input.config.sample_rate = mSampleRate;
    input.config.channel_mask = mChannelMask;
    input.config.format = mFormat;
    input.config.offload_info = mOffloadInfoCopy;
    input.clientInfo.clientUid = mClientUid;
    input.clientInfo.clientPid = mClientPid;
    input.clientInfo.clientTid = -1;
    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
        // It is currently meaningless to request SCHED_FIFO for a Java thread.  Even if the
        // application-level code follows all non-blocking design rules, the language runtime
        // doesn't also follow those rules, so the thread will not benefit overall.
        if (mAudioTrackThread != 0 && !mThreadCanCallJava) {
            input.clientInfo.clientTid = mAudioTrackThread->getTid();
        }
    }
    input.sharedBuffer = mSharedBuffer;
    input.notificationsPerBuffer = mNotificationsPerBufferReq;
    input.speed = 1.0;
    if (audio_has_proportional_frames(mFormat) && mSharedBuffer == 0 &&
            (mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
        input.speed  = !isPurePcmData_l() || isOffloadedOrDirect_l() ? 1.0f :
                        max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
    }
    input.flags = mFlags;
    input.frameCount = mReqFrameCount;
    input.notificationFrameCount = mNotificationFramesReq;
    input.selectedDeviceId = mSelectedDeviceId;
    input.sessionId = mSessionId;
    input.audioTrackCallback = mAudioTrackCallback;
    input.opPackageName = mOpPackageName;

    IAudioFlinger::CreateTrackOutput output;
   
// 调用AudioFlinger
    sp<IAudioTrack> track = audioFlinger->createTrack(input,
                                                      output,
                                                      &status);

    if (status != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
        ALOGE("%s(%d): AudioFlinger could not create track, status: %d output %d",
                __func__, mPortId, status, output.outputId);
        if (status == NO_ERROR) {
            status = NO_INIT;
        }
        goto exit;
    }
    ALOG_ASSERT(track != 0);
   // 利用flinger的返回重新设置参数
    mFrameCount = output.frameCount;
    mNotificationFramesAct = (uint32_t)output.notificationFrameCount;
    mRoutedDeviceId = output.selectedDeviceId;
    mSessionId = output.sessionId;

    mSampleRate = output.sampleRate;
    if (mOriginalSampleRate == 0) {
        mOriginalSampleRate = mSampleRate;
    }

    mAfFrameCount = output.afFrameCount;
    mAfSampleRate = output.afSampleRate;
    mAfLatency = output.afLatencyMs;

    mLatency = mAfLatency + (1000LL * mFrameCount) / mSampleRate;

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

    // FIXME compare to AudioRecord 
// 共享内存，这样应用可以和AudioFlinger可以以共享内存的方式传递音频了
    sp<IMemory> iMem = track->getCblk();
    if (iMem == 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).
    void *iMemPointer = iMem->unsecurePointer();

    mAudioTrack = track;
    mCblkMemory = iMem;
    IPCThreadState::self()->flushCommands();

    audio_track_cblk_t* cblk = static_cast<audio_track_cblk_t*>(iMemPointer);
    mCblk = cblk;

    mAwaitBoost = false;
    if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
        if (output.flags & AUDIO_OUTPUT_FLAG_FAST) {
            ALOGI("%s(%d): AUDIO_OUTPUT_FLAG_FAST successful; frameCount %zu -> %zu",
                  __func__, mPortId, mReqFrameCount, mFrameCount);
            if (!mThreadCanCallJava) {
                mAwaitBoost = true;
            }
        } else {
            ALOGD("%s(%d): AUDIO_OUTPUT_FLAG_FAST denied by server; frameCount %zu -> %zu",
                  __func__, mPortId, mReqFrameCount, mFrameCount);
        }
    }
    mFlags = output.flags;

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

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

    // Starting address of buffers in shared memory.  If there is a shared buffer, buffers
    // is the value of pointer() for the shared buffer, otherwise buffers points
    // immediately after the control block.  This address is for the mapping within client
    // address space.  AudioFlinger::TrackBase::mBuffer is for the server address space.
    void* buffers;
    if (mSharedBuffer == 0) {
        buffers = cblk + 1; // Stream 模式
    } 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 = mSharedBuffer->unsecurePointer();
        if (buffers == NULL) { // Static模式
            ALOGE("%s(%d): Could not get buffer pointer", __func__, mPortId);
            status = NO_INIT;
            goto exit;
        }
    }

    mAudioTrack->attachAuxEffect(mAuxEffectId);

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

    // reset server position to 0 as we have new cblk.
    mServer = 0;

    // update proxy
    if (mSharedBuffer == 0) { // Stream场景
        mStaticProxy.clear();
        mProxy = new AudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
    } else {// Static场景
        mStaticProxy = new StaticAudioTrackClientProxy(cblk, buffers, mFrameCount, mFrameSize);
        mProxy = mStaticProxy;
    }

    mProxy->setVolumeLR(gain_minifloat_pack(
            gain_from_float(mVolume[AUDIO_INTERLEAVE_LEFT]),
            gain_from_float(mVolume[AUDIO_INTERLEAVE_RIGHT])));

    mProxy->setSendLevel(mSendLevel);
    const uint32_t effectiveSampleRate = adjustSampleRate(mSampleRate, mPlaybackRate.mPitch);
    const float effectiveSpeed = adjustSpeed(mPlaybackRate.mSpeed, mPlaybackRate.mPitch);
    const float effectivePitch = adjustPitch(mPlaybackRate.mPitch);
    mProxy->setSampleRate(effectiveSampleRate);

    AudioPlaybackRate playbackRateTemp = mPlaybackRate;
    playbackRateTemp.mSpeed = effectiveSpeed;
    playbackRateTemp.mPitch = effectivePitch;
    mProxy->setPlaybackRate(playbackRateTemp);
    mProxy->setMinimum(mNotificationFramesAct);

    if (mDualMonoMode != AUDIO_DUAL_MONO_MODE_OFF) {
        setDualMonoMode_l(mDualMonoMode);
    }
    if (mAudioDescriptionMixLeveldB != -std::numeric_limits<float>::infinity()) {
        setAudioDescriptionMixLevel_l(mAudioDescriptionMixLeveldB);
    }

    mDeathNotifier = new DeathNotifier(this);
    IInterface::asBinder(mAudioTrack)->linkToDeath(mDeathNotifier, this);

    // This is the first log sent from the AudioTrack client.
    // The creation of the audio track by AudioFlinger (in the code above)
    // is the first log of the AudioTrack and must be present before
    // any AudioTrack client logs will be accepted.

    mMetricsId = std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(mPortId);
   
    return status;
}

sp<IAudioTrack> AudioFlinger::createTrack(const CreateTrackInput& input,
                                          CreateTrackOutput& output,
                                          status_t *status)
{
 ...
   //分配session id
    audio_session_t sessionId = input.sessionId;
    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.outputId = AUDIO_IO_HANDLE_NONE;
    output.selectedDeviceId = input.selectedDeviceId;
    lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
                                            clientPid, clientUid, &input.config, input.flags,
                                            &output.selectedDeviceId, &portId, &secondaryOutputs);

    if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
        ALOGE("createTrack() getOutputForAttr() return error %d or invalid output handle", lStatus);
        goto Exit;
    }
  ...
    { // 获取播放线程， 猜想2得到验证
        Mutex::Autolock _l(mLock);
        PlaybackThread *thread = checkPlaybackThread_l(output.outputId);
        if (thread == NULL) {
            ALOGE("no playback thread found for output handle %d", output.outputId);
            lStatus = BAD_VALUE;
            goto Exit;
        }

        client = registerPid(clientPid); // 创建共享内存，猜想3得到验证
    // 获取特效线程
        PlaybackThread *effectThread = NULL;
        // check if an effect chain with the same session ID is present on another
        // output thread and move it here.
        for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
            sp<PlaybackThread> t = mPlaybackThreads.valueAt(i);
            if (mPlaybackThreads.keyAt(i) != output.outputId) {
                uint32_t sessions = t->hasAudioSession(sessionId);
                if (sessions & ThreadBase::EFFECT_SESSION) {
                    effectThread = t.get();
                    break;
                }
            }
        }
        ALOGV("createTrack() sessionId: %d", sessionId);

        output.sampleRate = input.config.sample_rate;
        output.frameCount = input.frameCount;
        output.notificationFrameCount = input.notificationFrameCount;
        output.flags = input.flags;
 // 创建track，猜想1得到验证
        track = thread->createTrack_l(client, streamType, localAttr, &output.sampleRate,
                                      input.config.format, input.config.channel_mask,
                                      &output.frameCount, &output.notificationFrameCount,
                                      input.notificationsPerBuffer, input.speed,
                                      input.sharedBuffer, sessionId, &output.flags,
                                      callingPid, input.clientInfo.clientTid, clientUid,
                                      &lStatus, portId, input.audioTrackCallback,
                                      input.opPackageName);
        LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
        // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless

        output.afFrameCount = thread->frameCount();
        output.afSampleRate = thread->sampleRate();
        output.afLatencyMs = thread->latency();
        output.portId = portId;

        if (lStatus == NO_ERROR) {
            // Connect secondary outputs. Failure on a secondary output must not imped the primary
            // Any secondary output setup failure will lead to a desync between the AP and AF until
            // the track is destroyed.
            TeePatches teePatches;
            for (audio_io_handle_t secondaryOutput : secondaryOutputs) { //  候选输出
                PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
                if (secondaryThread == NULL) {
                    ALOGE("no playback thread found for secondary output %d", output.outputId);
                    continue;
                }

                size_t sourceFrameCount = thread->frameCount() * output.sampleRate
                                          / thread->sampleRate();
                size_t sinkFrameCount = secondaryThread->frameCount() * output.sampleRate
                                          / secondaryThread->sampleRate();
                // If the secondary output has just been opened, the first secondaryThread write
                // will not block as it will fill the empty startup buffer of the HAL,
                // so a second sink buffer needs to be ready for the immediate next blocking write.
                // Additionally, have a margin of one main thread buffer as the scheduling jitter
                // can reorder the writes (eg if thread A&B have the same write intervale,
                // the scheduler could schedule AB...BA)
                size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
                // Total secondary output buffer must be at least as the read frames plus
                // the margin of a few buffers on both sides in case the
                // threads scheduling has some jitter.
                // That value should not impact latency as the secondary track is started before
                // its buffer is full, see frameCountToBeReady.
                size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
                // The frameCount should also not be smaller than the secondary thread min frame
                // count
                size_t minFrameCount = AudioSystem::calculateMinFrameCount(
                            [&] { Mutex::Autolock _l(secondaryThread->mLock);
                                  return secondaryThread->latency_l(); }(),
                            secondaryThread->mNormalFrameCount,
                            secondaryThread->mSampleRate,
                            output.sampleRate,
                            input.speed);
                frameCount = std::max(frameCount, minFrameCount);

                using namespace std::chrono_literals;
                auto inChannelMask = audio_channel_mask_out_to_in(input.config.channel_mask);
                sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
                                                               output.sampleRate,
                                                               inChannelMask,
                                                               input.config.format,
                                                               frameCount,
                                                               NULL /* buffer */,
                                                               (size_t)0 /* bufferSize */,
                                                               AUDIO_INPUT_FLAG_DIRECT,
                                                               0ns /* timeout */);
                status_t status = patchRecord->initCheck();
                if (status != NO_ERROR) {
                    ALOGE("Secondary output patchRecord init failed: %d", status);
                    continue;
                }

                // TODO: We could check compatibility of the secondaryThread with the PatchTrack
                // for fast usage: thread has fast mixer, sample rate matches, etc.;
                // for now, we exclude fast tracks by removing the Fast flag.
                const audio_output_flags_t outputFlags =
                        (audio_output_flags_t)(output.flags & ~AUDIO_OUTPUT_FLAG_FAST);
                sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
                                                               streamType,
                                                               output.sampleRate,
                                                               input.config.channel_mask,
                                                               input.config.format,
                                                               frameCount,
                                                               patchRecord->buffer(),
                                                               patchRecord->bufferSize(),
                                                               outputFlags,
                                                               0ns /* timeout */,
                                                               frameCountToBeReady);
                status = patchTrack->initCheck();
                if (status != NO_ERROR) {
                    ALOGE("Secondary output patchTrack init failed: %d", status);
                    continue;
                }
                teePatches.push_back({patchRecord, patchTrack});
                secondaryThread->addPatchTrack(patchTrack);
                // In case the downstream patchTrack on the secondaryThread temporarily outlives
                // our created track, ensure the corresponding patchRecord is still alive.
                patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
                patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
            }
            track->setTeePatches(std::move(teePatches));
        }

        // move effect chain to this output thread if an effect on same session was waiting
        // for a track to be created
        if (lStatus == NO_ERROR && effectThread != NULL) {
            // no risk of deadlock because AudioFlinger::mLock is held
            Mutex::Autolock _dl(thread->mLock);
            Mutex::Autolock _sl(effectThread->mLock);
            if (moveEffectChain_l(sessionId, effectThread, thread) == NO_ERROR) {
                effectThreadId = thread->id();
                effectIds = thread->getEffectIds_l(sessionId);
            }
        }

        // Look for sync events awaiting for a session to be used.
        for (size_t i = 0; i < mPendingSyncEvents.size(); i++) {
            if (mPendingSyncEvents[i]->triggerSession() == sessionId) {
                if (thread->isValidSyncEvent(mPendingSyncEvents[i])) {
                    if (lStatus == NO_ERROR) {
                        (void) track->setSyncEvent(mPendingSyncEvents[i]);
                    } else {
                        mPendingSyncEvents[i]->cancel();
                    }
                    mPendingSyncEvents.removeAt(i);
                    i--;
                }
            }
        }
        if ((output.flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) == AUDIO_OUTPUT_FLAG_HW_AV_SYNC) {
            setAudioHwSyncForSession_l(thread, sessionId);
        }
    }

    if (lStatus != NO_ERROR) {
        // remove local strong reference to Client before deleting the Track 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();
        }
        track.clear();
        goto Exit;
    }

    // effectThreadId is not NONE if an effect chain corresponding to the track session
    // was found on another thread and must be moved on this thread
    if (effectThreadId != AUDIO_IO_HANDLE_NONE) {
        AudioSystem::moveEffectsToIo(effectIds, effectThreadId);
    }

    // return handle to client
    trackHandle = new TrackHandle(track); // 验证猜想1， 将track包装成binder

Exit:
    if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
        AudioSystem::releaseOutput(portId);
    }
    *status = lStatus;
    return trackHandle;
}

// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
    return mPlaybackThreads.valueFor(output).get();
}

        MIXER,              // Thread class is MixerThread
        DIRECT,             // Thread class is DirectOutputThread
        DUPLICATING,        // Thread class is DuplicatingThread
        RECORD,             // Thread class is RecordThread
        OFFLOAD,            // Thread class is OffloadThread
        MMAP_PLAYBACK,      // Thread class for MMAP playback stream
        MMAP_CAPTURE,       // Thread class for MMAP capture stream

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());
}

size_t AudioFlinger::getClientSharedHeapSize() const
{
    size_t heapSizeInBytes = property_get_int32("ro.af.client_heap_size_kbyte", 0) * 1024;
    if (heapSizeInBytes != 0) { // read-only property overrides all.
        return heapSizeInBytes;
    }
    return mClientSharedHeapSize;
}

sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
        const sp<AudioFlinger::Client>& client,
        audio_stream_type_t streamType,
        const audio_attributes_t& attr,
        uint32_t *pSampleRate,
        audio_format_t format,
        audio_channel_mask_t channelMask,
        size_t *pFrameCount,
        size_t *pNotificationFrameCount,
        uint32_t notificationsPerBuffer,
        float speed,
        const sp<IMemory>& sharedBuffer,
        audio_session_t sessionId,
        audio_output_flags_t *flags,
        pid_t creatorPid,
        pid_t tid,
        uid_t uid,
        status_t *status,
        audio_port_handle_t portId,
        const sp<media::IAudioTrackCallback>& callback,
        const std::string& opPackageName)
{
...
       track = new Track(this, client, streamType, attr, sampleRate, format,
                          channelMask, frameCount,
                          nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
                          sessionId, creatorPid, uid, *flags, TrackBase::TYPE_DEFAULT, portId,
                          SIZE_MAX /*frameCountToBeReady*/, opPackageName);

        lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
        if (lStatus != NO_ERROR) {
            ALOGE("createTrack_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);
        {
            Mutex::Autolock _atCbL(mAudioTrackCbLock);
            if (callback.get() != nullptr) {
                mAudioTrackCallbacks.emplace(track, callback);
            }
        }

        sp<EffectChain> chain = getEffectChain_l(sessionId);
        if (chain != 0) {
            ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
            track->setMainBuffer(chain->inBuffer());
            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
            chain->incTrackCnt();
        }
...
}

// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
            PlaybackThread *thread,
            const sp<Client>& client,
            audio_stream_type_t streamType,
            const audio_attributes_t& attr,
            uint32_t sampleRate,
            audio_format_t format,
            audio_channel_mask_t channelMask,
            size_t frameCount,
            void *buffer,
            size_t bufferSize,
            const sp<IMemory>& sharedBuffer,
            audio_session_t sessionId,
            pid_t creatorPid,
            uid_t uid,
            audio_output_flags_t flags,
            track_type type,
            audio_port_handle_t portId,
            size_t frameCountToBeReady,
            const std::string opPackageName)
    :   TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
                  // 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).
                  (sharedBuffer != 0) ? sharedBuffer->unsecurePointer() : buffer,
                  (sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
                  sessionId, creatorPid, uid, true /*isOut*/,
                  (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
                  type,
                  portId,
                  std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_TRACK) + std::to_string(portId)),
    mFillingUpStatus(FS_INVALID),
    // mRetryCount initialized later when needed
    mSharedBuffer(sharedBuffer),
    mStreamType(streamType),
    mMainBuffer(thread->sinkBuffer()),
    mAuxBuffer(NULL),
    mAuxEffectId(0), mHasVolumeController(false),
    mFrameMap(16 /* sink-frame-to-track-frame map memory */),
    mVolumeHandler(new media::VolumeHandler(sampleRate)),
    mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(
            uid, attr, id(), streamType, opPackageName)),
    // mSinkTimestamp
    mFastIndex(-1),
    mCachedVolume(1.0),
    /* The track might not play immediately after being active, similarly as if its volume was 0.
     * When the track starts playing, its volume will be computed. */
    mFinalVolume(0.f),
    mResumeToStopping(false),
    mFlushHwPending(false),
    mFlags(flags)
{

    if (sharedBuffer == 0) {
        mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount, // Stream模式，把共享内存地址传递给proxy
                mFrameSize, !isExternalTrack(), sampleRate);
    } else {
        mAudioTrackServerProxy = new StaticAudioTrackServerProxy(mCblk, mBuffer, frameCount, // STATIC 模式，把共享内存地址传递给proxy
                mFrameSize, sampleRate);
    }
    mServerProxy = mAudioTrackServerProxy; // 到了这儿就可以通过mServerProxy获取共享内存了
    mServerProxy->setStartThresholdInFrames(frameCountToBeReady); // update the Cblk value


    if (channelMask & AUDIO_CHANNEL_HAPTIC_ALL) {
        mAudioVibrationController = new AudioVibrationController(this);
        mExternalVibration = new os::ExternalVibration(
                mUid, opPackageName, mAttr, mAudioVibrationController);
    }
}

// TrackBase constructor must be called with AudioFlinger::mLock held
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)
{

    if (client != 0) {
        mCblkMemory = client->heap()->allocate(size);  // 从共享内存中分配内存
        if (mCblkMemory == 0 ||
                (mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->unsecurePointer())) == NULL) { // mCblk 记录了内存地址
            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

    }
}

AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track)
    : BnAudioTrack(),
      mTrack(track)
{
}

AudioFlinger::TrackHandle::~TrackHandle() {
    // just stop the track on deletion, associated resources
    // will be freed from the main thread once all pending buffers have
    // been played. Unless it's not in the active track list, in which
    // case we free everything now...
    mTrack->destroy();
}

sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
    return mTrack->getCblk();
}

status_t AudioFlinger::TrackHandle::start() {
    return mTrack->start();
}

void AudioFlinger::TrackHandle::stop() {
    mTrack->stop();
}

void AudioFlinger::TrackHandle::flush() {
    mTrack->flush();
}

void AudioFlinger::TrackHandle::pause() {
    mTrack->pause();
}