[alsa-devel] What does snd_pcm_delay() actually return?

[Jaroslav Kysela]

On Fri, 13 Jun 2008, Takashi Iwai wrote:

> At Fri, 13 Jun 2008 18:47:48 +0200 (CEST),
> Jaroslav Kysela wrote:
> > 
> > On Fri, 13 Jun 2008, Takashi Iwai wrote:
> > 
> > > At Fri, 13 Jun 2008 18:11:12 +0200 (CEST),
> > > Jaroslav Kysela wrote:
> > > > 
> > > > On Fri, 13 Jun 2008, Takashi Iwai wrote:
> > > > 
> > > > > What about just providing three pointers: curr_ptr, hw_ptr and
> > > > > appl_ptr?  curr_ptr corresponds to the point being played, and hw_ptr
> > > > > is the point where the data was already sent to h/w, and appl_ptr is
> > > > > the point where the data is filled by user.  The above definitions are
> > > > > all combinations of these pointers.
> > > > 
> > > > But I think that curr_ptr can be managed in drivers, thus invisible to 
> > > > user space (except for snd_pcm_delay() propagation).
> > > 
> > > Ditto for hw_ptr.  Why is it hidden at all?
> > 
> > Does it improve something to show this pointer to apps? I don't see any 
> > reason to show it outside alsa-lib.
> 
> Then it'll be more clear.

Maybe for us, but not for application developers. They need to know only 
how much samples are available for I/O operation.

> > > > If driver requires 
> > > > extra handling of samples, it can allocate and manage extra buffers 
> > > > itself. I don't see the point to have "locked" samples already processed 
> > > > by hardware in the main ring buffer described by appl_ptr / hw_ptr.
> > > > Application can use this space for new samples.
> > > > 
> > > > The only advantage with your implementation might be zero-copy, but USB 
> > > > and PCMCIA cards have or create own buffers, so I don't think that this 
> > > > advantage can be used in actual drivers and I cannot even imagine 
> > > > hardware which work in way to use zero-copy in this situation.
> > > 
> > > Wait, wait.  Please don't mix up.  The above doesn't imply anything
> > > about the further implementation of usb-audio driver.  What I
> > > suggested is, instead of hiding two pointers (hw_ptr and curr_ptr) and
> > > creating a complex API, simply expose them.
> > 
> > I don't see a reason to make current API more complex.
> 
> Because the current API is complex and hard to understand.

But this concrete part of API is quite simple, isn't?

> > We have already two 
> > functions, One showing overall latency and second one how much samples can 
> > be processed by application. It's enough. We need only improve things 
> > internaly in alsa-lib <-> kernel (provide correct information for 
> > snd_pcm_delay()).
> > 
> > > Now, regarding the usb-driver.  Honestly, I don't understand what you
> > > want to do with an extra URB.
> > 
> > Note that we don't need to have extra URBs, just change hw_ptr handling 
> > in USB driver.
> 
> OK, then it's different from your previous explanation...

Yes, sorry for not perfect explanation. I meant this.

> > > As now, usb-audio driver handles as curr_ptr == hw_ptr.  But, in
> > > reality, curr_ptr = hw_ptr - samples_in_urbs.  So, in the case
> > > of USB-audio, hw_ptr is ahead of curr_ptr.  (And the granularity is
> > > samples_in_urbs).
> > 
> > As Lennart mentioned, in this case you can reach underrun at different 
> > position than expected (when URB cannot be filled). In my case, you'll 
> > reach underrun exactly at point when whole ring buffer is drained. So 
> > application can better estimate queueing and also it makes things more 
> > logical.
> 
> Hm, could you elaborate how to do this more exactly?  That wasn't
> clear from your previous post at all.

Looking to USB driver, snd_period_elapsed() is called directly after URB 
is filled (of course when it crosses the period boundary). Also 
hwptr_done variable is updated in this time.

It means that the PCM midlevel code thinks that samples in URBs are 
played (underrun can be detected), but they are queued in URBs.

OK, my fault. It's exactly behaviour I proposed (URBs are extra buffers), 
but we need to take in account the right snd_pcm_delay() output. Lennart 
probably meant that samples are consumed too much quickly at the stream 
start and impossibility to detect the extra buffering mechanism with the 
current code.

I would propose to add an extra callback to snd_pcm_ops to determine 
queued samples by driver and/or in hardware and extend snd_pcm_status and 
snd_pcm_mmap_status structures to propagate this value to user space.

					Jaroslav

-----
Jaroslav Kysela <perex at perex.cz>
Linux Kernel Sound Maintainer
ALSA Project, Red Hat, Inc.