> Simply stated, I can drop a DS Audio Capture filter and render
> directly to a DirectSound render device in GraphEdit with a 1/2 second
> delay. If in the same GraphEdit, I use the WDM streaming audio source
> for my mic and attach to the same DirectSound renderer there is very
> little delay. So it appears to me that the DS audio capture filter is
> "adding" about 1/2 second delay under Vista. The same seems to be
> happening using DirectSound technology. ANy ideas? I'm trying to
> write an VoIP application and I'm getting that stupid 1/2 second delay
> right out of the chute which is unacceptable.

> On Thu, 21 Aug 2008 09:48:26 -0700 (PDT), Loren Rogers wrote:

>> Simply stated, I can drop a DS Audio Capture filter and render
>> directly to a DirectSound render device in GraphEdit with a 1/2 second
>> delay.  If in the same GraphEdit, I use the WDM streaming audio source
>> for my mic and attach to the same DirectSound renderer there is very
>> little delay.  So it appears to me that the DS audio capture filter is
>> "adding" about 1/2 second delay under Vista.  The same seems to be
>> happening using DirectSound technology.  ANy ideas?  I'm trying to
>> write an VoIP application and I'm getting that stupid 1/2 second delay
>> right out of the chute which is unacceptable.

>
> It's the default buffer size.  The default...

> Is there anything I can do to rectify this? My goal is to have 20ms
> worth of samples delivered from the capture source.

> On Thu, 21 Aug 2008 14:37:06 -0700 (PDT), Loren Rogers wrote:

>> Is there anything I can do to rectify this?  My goal is to have 20ms
>> worth of samples delivered from the capture source.

>
> You should be able to use IAMBufferNegotiation at the output pin of the
> capture source before connecting it to the renderer.
>
> For some filters you have to use an intermediate filter to force the buffer
> during the connection process but you shouldn't have to here.
>
> --http://www.chrisnet.net/code.htm
> [MS MVP for DirectShow / MediaFoundation]

> Ya I see and you are right! However my "real" configuration is this,
> and I would really appreciate if you could give me some guidance of
> what I'm doing wrong because I'm still getting a delay in...

> On Thu, 21 Aug 2008 19:17:21 -0700 (PDT), Loren Rogers wrote:

>> Ya I see and you are right!  However my "real" configuration is this,
>> and I would really appreciate if you could give me some guidance of
>> what I'm doing wrong because I'm still getting a delay in my
>> configuration.
>> Graph1: Capture (20ms samples 16KHz, 16 bits) to SampleGrabber, then
>> sample grabber to Null renderer.  Null renderer is optional
>> Graph2: I created a custom CSource/CSourceStream "push" filter to push
>> media samples to default DirectSound renderer filter(speakers).
>> CSourceStream::DecideBufferSize() sets up the same params as I use in
>> the source's IAMBufferNegotiation
>> I "grab" 20 ms samples from SampleGrabber, and push the sample(s) onto
>> a queue.
>> Then, the CSourceStream::FillBuffer() (which is obviously looping),
>> then takes those samples and fills the IMediaSample passed to it.
>> I'm obviously doing something wrong here, and if you could point me in
>> the right direction, I'd appreciate it.

>
> Part 2 of the problem is that the DirectSound renderer has an internal 1 ...

> I implemented the IAMPushSource on my output pin (CSourceStream) and I
> notice that the only
> method that get's called is GetPushSourceFlags, and I've attempted to
> set that flag (returned) to a different value all with the same
> results... an audio delay