On Wed, 21.01.09 01:39, Takashi Iwai (tiwai@xxxxxxx) wrote:
> > The function should look like this:
> >
> > snd_pcm_sframes_t snd_pcm_busy_for(snd_pcm_t *pcm);
> >
> > I called the prototype "busy for" since effectively the value I am
> > looking for is the time the card will be busy with the data it already
> > has, and doesn't need any new data.
>
> Isn't it snd_pcm_delay() that was originally designed for?
No. Let me summarize the meaning of snd_pcm_update_avail(),
snd_pcm_delay() and my snd_pcm_busy_for() to opefully make clear where
the differences are:
snd_pcm_update_avail() -- returns how many samples can be written
right now without blocking.
snd_pcm_delay() -- returns how many samples will be played before the
samples that are written now can be heard.
snd_pcm_busy_for() -- returns how many samples will be played before
ALSA would enter an underrun situation if no
further samples are written.
snd_pcm_update_avail() and snd_pcm_busy_for() return metrics that are
solely dependant on the size and metrics of the hardware buffer and
its current indexes. snd_pcm_delay() also includes information about
any extra latency that comes after the playback buffer.
Onle snd_pcm_update_avail()/snd_pcm_busy_for() are influenced by "fast
starts" as done by the USB driver's double buffering and by
block-based transfer.
Hmm, I am trying my best to explain why I want this function and what
exactly it should do. Any chance I can convince you guys that this
function really matters for timer-based audio scheduling?
> Did you check my previous patch?
You mean the one that makes snd_pcm_delay() for USB devices actually
include the extra latency that comes after the playback buffer? No, I
didn't check that one yet. It takes so much time to patch the kernel
and test things... I'll try too finally do it this WE.
Lennart
--
Lennart Poettering Red Hat, Inc.
lennart [at] poettering [dot] net ICQ# 11060553
http://0pointer.net/lennart/ GnuPG 0x1A015CC4
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