Hope this helps
Arve Knudsen
On Thu, 06 May 2004 18:15:56 +0000, James Courtier-Dutton <[EMAIL PROTECTED]> wrote:
Hi,
I have been updating the wine alsa driver to work better with alsa.
So far, all I have done is update it to use the new alsa api.
Windows uses an api called Direct Sound.
Direct Sound uses direct hardware buffer access.
A Win32 program can quiry the sound driver and ask for the currently playing position, together with the first available possition that the application could write to. I will call these "PlayPos" and "WritePos" respectively.
E.g. If the hardware is halfway through playing one period(the PlayPos is in the middle of a period), the beginning of the next period's position is returned as the "WritePos".
The problem arrives because the Win32 app expects to have full write access to the hardware buffer, and can write to any parts of it at any time. The sound card should just happily move through the buffer playing each sample as it goes, and when it reaches the end, just start at the beginning again.
The alsa api is based more around the sound card asking for the next block of samples.
Strangely enough, it Direct Sound works fine when using oss emulation in alsa.
I was trying to just get alsa to copy samples from the Direct Sound buffer on a just in time basis.
I was starting with writing one period to the sound card, calling snd_pcm_start(), then setting snd_async_add_pcm_handler(), expecting the handler to be called on each period, but the handler is never called.
For some reason, the handler does not seem to get called on each period elasped as I was expecting.
Another problem is that if I want to stop the stream, I call "snd_pcm_drop(). With this I would expect snd_pcm_delay()==0 and snd_pcm_avail_update()==buffer_size.
It seems that these values are only reset to 0 after first calling snd_pcm_prepare().
After stopping the stream in this way, I want to restart it again without having to close/open the pcm.
Can anyone give me advice as to what I am doing wrong ?
Cheers James
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