I have a Delphi 6 Pro app that uses the DSPACK component library to send audio to Skype from the system's preferred audio input device. I am using a TSampleGrabber component to tap into the Filter Graph chain and then send the audio buffers to Skype. The problem is that I am only getting audio once a second. In other words, the OnBuffer() event for the TSampleGrabber instance only fires once a second with a full second's worth of data in the Buffer parameter. I need to know how to modify my Filter Graph chain so it grabs data from the input device at a faster interval than once a second. If possible, I'd like to do it as fast as every 50 ms or at least every 100ms.
My Filter Graph chain consists of a TFilter that is mapped to the system's preferred audio input device at the top. I attach the output pins of that filter to the input pins of a 'WAV Dest' assigned TFilter so I can get the samples in PCM WAV format. I then attach the output pins of the 'WAV Dest' filter to the input pins of the TSampleGrabber instance. What do I need to change to get the TSampleGrabber OnBuffer() event to fire at a faster interval?
UPDATE: Based on Roman R's answer I was able to implement a solution that I am showing below. One note. His link led me to the following blog post that was helpful in solution:
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;
