Re: sound rendering problem

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Hmmm. Very interesting information.
Because MSVAD is software device, i suppose that internal clock - timer that
triggers with 10ms interval.
To tell the truth, i don't quite understand what request is responsible for
audio data moving.
Is it Dpc that triggers with timer or IMiniportWavePciStream::GetPosition
client request? There are only one call IPortWaveCyclic::Notify in dpc.
Using debug output i see calls for IMiniportWavePciStream::GetPosition that
calculates byte dispacements and subsequent CopyFrom calls for total length
= bytedispacement.

source data will coming from another part of MSVAD. i planed use mvsad audio
renderer filter for passing custom pcm data within scope of directshow. so
source data in general may coming from another live audio source, file with
audio stream or custom audio source filter.
currently i simple use preloaded audio buffer from the disk and tesing audio
playback.

"Chris P." <msdn@xxxxxxxxxxxx> wrote in message
news:1u5m56u9yv40a.k14l6pxennld.dlg@xxxxxxxxxxxxx
On Tue, 23 Jan 2007 18:01:53 +0600, Tulup wrote:

Implementing IReferenceClock could be difficult as you are being wrapped
by
a DirectShow filter.

By default audio capture sources do not implement the clock because the
general assumption was made that the source would be using the same clock
as the renderer, i.e. the source was the renderer device. You'll notice
that the exact same problem occurs if you take the source from one sound
card and feed it to the renderer of another.

The problem is that each device has a different internal clock, this
difference makes it more or less impossible to synchronize streams
perfectly, you can only give the renderer hints on how to best
synchronize.

Perhaps if you give a little more info on where your source data is coming
from we can make other suggestions?

--
http://www.chrisnet.net/code.htm
http://www.avdevforum.com/AV/viewforum.php?f=7


.

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