如何消除DirectShow筛选器链中的1秒延迟？（使用Delphi和DSPACK）

我有一个Delphi 6 Pro应用程序，它使用DSPACK组件库从系统首选的音频输入设备向Skype发送音频。我正在使用TSampleGrabber组件点击过滤图链，然后将音频缓冲区发送到Skype。问题是我每秒只收到一次音频。换句话说，TsSampleGrabber实例的OnBuffer（）事件每秒只触发一次，Buffer参数中有一整秒钟的数据。我需要知道如何修改过滤图链，以便它以比每秒一次更快的间隔从输入设备获取数据。如果可能的话，我希望每50毫秒或至少每100毫秒做一次

我的过滤图链由一个映射到顶部系统首选音频输入设备的TFilter组成。我将该过滤器的输出管脚连接到指定TFilter的“WAV Dest”输入管脚，以便以PCM WAV格式获取样本。然后，我将“WAV Dest”过滤器的输出管脚连接到TSampleGrabber实例的输入管脚。要以更快的间隔触发TSampleGrabber OnBuffer（）事件，我需要做哪些更改

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更新：根据Roman R的回答，我能够实现我在下面展示的解决方案。一个音符。他的链接让我看到了以下对解决方案很有帮助的博客文章：

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我想您需要微调音频捕获过滤器，以在缓冲区中捕获您想要的大小，即足够短以使总延迟变小

音频捕获过滤器在输出管脚上公开

iambufference

接口，并且

SuggestAllocatorProperties

允许您指定缓冲区配置

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有关更多信息，请参阅：。

谢谢@Roman R。我已更新了我的原始帖子，加入了我在您的原始链接后找到的解决方案。

// Variable declaration for output pin to manipulate.
var
    intfCapturePin: IPin;

...............


    // Put this code after you have initialized your audio capture device
    //  TFilter instance *and* set it's wave audio format.  My variable for
    //  this is FFiltAudCap.  I believe you need to set the buffer size before
    //  connecting up the pins of the Filters.  The media type was
    //  retrieved earlier (theMediaType) when I initialized the audio
    //  input device Filter so you will need to do similarly.

    // Get a reference to the desired output pin for the audio capture device.
    with FFiltAudCap as IBaseFilter do
        CheckDSError(findPin(StringToOleStr('Capture'), intfCapturePin));

    if not Assigned(intfCapturePin) then
        raise Exception.Create('Unable to find the audio input device''s Capture output pin.');

    // Set the capture device buffer to 50 ms worth of audio data to
    //  reduce latency.  NOTE: This will fail if the device does not
    //  support the latency you desire so make sure you watch out for that.
    setBufferLatency(intfCapturePin as IAMBufferNegotiation, 50, theMediaType);

..................

// The setBufferLatency() procedure.
procedure setBufferLatency(
                // A buffer negotiation interface pointer.
                intfBufNegotiate: IAMBufferNegotiation;
                // The desired latency in milliseconds.
                bufLatencyMS: WORD;
                // The media type the audio stream is set to.
                theMediaType: TMediaType);
var
    allocProp: _AllocatorProperties;
    wfex: TWaveFormatEx;
begin
    if not Assigned(intfBufNegotiate) then
        raise Exception.Create('The buffer negotiation interface object is unassigned.');

    // Calculate the number of bytes per second using the wave
    // format belonging to the given Media Type.
    wfex := getWaveFormat(theMediaType);

    if wfex.nAvgBytesPerSec = 0 then
        raise Exception.Create('The average bytes per second value for the given Media Type is 0.');

    allocProp.cbAlign := -1;  // -1 means "no preference".
    // Calculate the size of the buffer needed to get the desired
    //  latency in milliseconds given the average bytes per second
    //  of the Media Type's audio format.
    allocProp.cbBuffer := Trunc(wfex.nAvgBytesPerSec * (bufLatencyMS / 1000));
    allocProp.cbPrefix := -1;
    allocProp.cBuffers := -1;

    // Try to set the buffer size to the desired.
    CheckDSError(intfBufNegotiate.SuggestAllocatorProperties(allocProp));
end;