[vorbis-dev] Skipping with vorbisfile playback using DirectSound

[Brett Paterson]

Fmod sound system uses 200ms (x2 doublebuffer) for streaming and ogg
plays back pretty smooth.  It will skip if the media is slow though, and
1000-1500ms buffersizes can be needed from CD for example.

 
Brett Paterson
CEO / Lead Programmer.
Firelight Technologies Pty Ltd.
FMOD SoundSystem
http://www.fmod.org
 

> -----Original Message-----
> From: owner-vorbis-dev at xiph.org [mailto:owner-vorbis-dev at xiph.org] On
> Behalf Of Borut Pfeifer
> Sent: Thursday, 12 September 2002 1:17 AM
> To: vorbis-dev at xiph.org
> Subject: Re: [vorbis-dev] Skipping with vorbisfile playback using
> DirectSound
> 
> I'm ahead of you on that one - I already use
DSBCAPS_GETCURRENTPOSITION2
> (although the possibility exists that I'm writing to the wrong place
when
> streaming in .ogg files, but not .wavs, which I will look into).  What
> size
> buffer did you use?
> 
>    Borut
> 
> ----- Original Message -----
> From: "Ryan Wellman" <firestar at eskimo.com>
> To: <vorbis-dev at xiph.org>
> Sent: Tuesday, September 10, 2002 9:11 PM
> Subject: Re: [vorbis-dev] Skipping with vorbisfile playback using
> DirectSound
> 
> 
> > When I first tried implementing something like this I had a similar
> > problem (some sound cards it would happen with more than others).
In
> any
> > case it came down to the fact that play cursor was getting reported
in
> odd
> > (and highly inaccurate) locations.  Adding the following flags when
> > creating the bufferdesc DSBCAPS_GETCURRENTPOSITION2 moved it closer
to
> > where I was expecting it.  Although we did end up dividing our
buffer
> into
> > thirds just to be safe. :)
> >
> > Ryan
> >
> > On Tue, 10 Sep 2002, Borut Pfeifer wrote:
> >
> > > I'm having a problem trying to write a simple vorbis file player
using
> > > DirectSound. The decoding portion is pretty much straight from the
> > > vorbisfile sample code.  The pcm data gets put into one half of a
> > > DirectSound buffer, and as that half is playing, the next half
gets
> filled.
> > > The problem is, there's skipping and some noise when the file is
being
> > > played (although some of the music is audible). Could this this
caused
> by
> > > the time spent decoding? Wav files work fine so I'm fairly certain
> it's
> not
> > > the playback code itself. I've tried decoding various amounts at
one
> time
> > > but in each case the same thing occurs (although to different
> degrees).
> Is
> > > there something I can do to fix this without adding a separate
> decoding
> > > thread (or is that going to be necessary)? Or could it be
something
> else?
> > >
> > > Thanks,
> > >   Borut
> > >
> > > --- >8 ----
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