[alsa-devel] Trouble understanding ALSA's DMA buffers

[Pharaoh .]

On 6/12/07, Takashi Iwai <tiwai at suse.de> wrote:
>
> At Tue, 12 Jun 2007 01:59:45 +0530,
> Nobin Mathew wrote:
> >
> > Hi,
> >
> > Now ALSA (audio ) buffer is divided into periods, i.e. a chain of small
> packets.
> >
> > periods size is configurable. Data transfer to the codec starts only
> > after reaching start_threshold point (start_threshold is in periods),
> > this time DMA trigger is called.
> >
> > This trigger onwards application will get notification from the kernel
> > saying that period buffer is empty you can write into it.Till the end
> > of music file.
>
> Yes.  And the "ping-poing" is the case that you have two periods in a
> ring buffer.
>
> Most hardwares support more periods practically.  That's why "periods"
> (corresponds to "fragments" in OSS) was introduced, as more generic
> abstraction.
>
>
> Takashi
>
> >
> > Nobin Mathew
> >
> >
> >
> > On 6/12/07, Timur Tabi <timur at freescale.com> wrote:
> > > I'm writing an ALSA SOC driver for an I2S-based device, and I'm having
> a really hard time
> > > understanding how ALSA uses the DMA buffers.  And yes, I've read the
> documentation and
> > > studied some sample source code.
> > >
> > > I used to write audio drivers for a living, but that was many years
> ago, and it wasn't for
> > > Linux.  Perhaps the concepts in my head are outdated, but I just don't
> see enough
> > > explanation as to how DMA buffers are supposed to work.
> > >
> > > Back then, audio drivers used "ping pong" DMA buffers.  A single DMA
> buffer is allocated,
> > > and the audio hardware is programmed to read from that buffer in a
> loop.  That is, it
> > > would automatically restart reading from the beginning of the buffer
> without any
> > > reprogramming.  The hardware would also be programmed to issue an
> interrupt when it got to
> > > the end of the buffer, and when it got to the half-way point.
> > >
> > > To start playback, you first filled the whole buffer with audio data,
> and then told the
> > > hardware to start playing.  After the hardware got to the half-way
> point, it would issue
> > > an interrupt.  You would then tell the OS you need more data, and
> you'd get it.  You then
> > > copy that data into the first half of the buffer *while* the hardware
> was playing the
> > > second half.  Later, the hardware would interrupt you when it got to
> the end of the
> > > buffer.  You'd then copy more data to the 2nd half while the hardware
> is playing the first
> > > half.
> > >
> > > And so - hardware plays one half while you copy data to the other
> half.  Hence, "ping pong".
> > >
> > > So how do I implement this in ALSA?  The "Writing an ALSA Driver"
> document doesn't even
> > > contain the words "ping" or "pong".
> > >
> > > --
> > > Timur Tabi
> > > Linux Kernel Developer @ Freescale
> > > _______________________________________________
> > > Alsa-devel mailing list
> > > Alsa-devel at alsa-project.org
> > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
> > >
> > _______________________________________________
> > Alsa-devel mailing list
> > Alsa-devel at alsa-project.org
> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
> >
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>


Hi

yet another newbie question about periods here:

1. AFAIK, the period size is closely dependent on the h/w, but after reading
some docs I collected
that,  they can be given values depending on how much we care about the
latency. Does it mean
that, I can vary it without paying any attention to what h/w manual says
just because I want low or
high latency? I hope this question is clear.

2. As periods correspond to fragment size from OSS world, what the other
periods related fields
correspond to i.e. what do the following fields mean?

period_bytes_min,
period_bytes_max,
periods_min,
periods_max,

I know what they mean after looking at them but I want to know the
relationship between various
fields.

For e.g.

I have,

buffer_bytes_max = 8192 * 8
i.e. = AUDIO_FRAGSIZE_DEFAULT * AUDIO_NBFRAGS_DEFAULT

here AUDIO_FRAGSIZE_DEFAULT is size of period right? Then to get the max
buffer size we should multiply it
by number of periods, is this correct? Also, these are default values of the
period and no of periods, then do
I need to see the h/w manual to decide the periods_min/periods_max and
period_bytes_min/period_bytes_max
fields?