我有一个Delphi 6 Pro应用程序,它使用DSPACK组件库从系统的首选音频输入设备发送音频到Skype。我使用一个TSampleGrabber组件来点击过滤器图形链,然后发送音频缓冲区到Skype。问题是,我一秒钟只收到一次音频。换句话说,针对OnBuffer实例的TSampleGrabber ()事件每秒钟只触发一次,缓冲区参数中包含完整秒的数据。我需要知道如何修改我的过滤器图形链,以便它以比每秒一次更快的间隔从输入设备抓取数据。如果可能的话,我想每隔50毫秒或至少每100毫秒做一次。
我的过滤器图形链由一个TFilter组成,它被映射到系统顶部的首选音频输入设备。我将该过滤器的输出引脚附加到分配给TFilter的“WAV”输入引脚上,这样我就可以获得PCM格式的样本。然后,我将“WAV”过滤器的输出引脚附加到TSampleGrabber实例的输入引脚上。要使TSampleGrabber OnBuffer()事件以更快的间隔触发,需要更改什么?
UPDATE:基于Roman的答案,我实现了一个解决方案,如下所示。一个音符。他的链接使我找到了以下有助于解决方案的博客文章:
http://sid6581.wordpress.com/2006/10/09/minimizing-audio-capture-latency-in-directshow/
// 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;发布于 2011-10-24 20:53:45
我认为您需要微调音频捕获过滤器,以捕获您想要的缓冲区,即足够短,以使整个延迟小。
音频捕获过滤器在输出引脚上公开IAMBufferNegotiation接口,SuggestAllocatorProperties允许您指定缓冲区配置。
更多信息见:Configuring Windows Media Audio Encoder DMO to reduce delay。
https://stackoverflow.com/questions/7881420
复制相似问题