DirectSound and i810 soundcards

[Ove Kaaven]

On Sat, 28 Dec 2002, Francois Gouget wrote:

> When DSound initializes it create a primary sound buffer and invokes 
> winmm to actually get access to a sound device. The default format of
> the primary buffer is 22050x16x2 (i.e. 22050Hz, 16 bit sound, stereo). 
> This means the device it's going to get will be handled by msacm.drv 
> which will be responsible for performing the conversions on the fly.
> 
> However during the primary buffer creation DSDB_MapPrimary gets invoked 
> in wineoss and does an mmap of the hardware sound buffer.
> 
> So when we then play sound in a secondary sound buffer, this sound gets 
> mixed into the primary buffer, but because of the mmap this then goes 
> straight to the sound card with no further conversion. So you end up 
> with a chipmunk sound effect since you play a 22050x16x2 sound at 
> 48000x16x2.
> 
> I will try to make more tests to confirm this but does it seem to make 
> sense? Should I look in some other direction? Any suggestion about a fix?

I've been meaning to submit the winex wineoss modifications, but haven't
got around to it yet, and it might be somewhat complicated by all the
changes around it lately. But this particular issue doesn't take a big
patch. Basically the 'true' sample rate should be propagated back to
dsound, so that it knows that the primary buffer is 48kHz and can resample
secondary buffers accordingly. No changes is needed to dsound for this,
wineoss's wodOpen just need to modify its waveformat input parameter if
the WAVE_DIRECTSOUND flag is present, then dsound will happily convert
secondary buffers to that sample rate when mixing them into the primary
buffer. (Primary buffers are different, but apps that writes directly into
the primary buffer are few, and they should use GetFormat and resample on
their own.)