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Message-ID: <20160620114924.GA8971@sisyphus.home.austad.us>
Date: Mon, 20 Jun 2016 13:49:24 +0200
From: Henrik Austad <henrik@...tad.us>
To: Pierre-Louis Bossart <pierre-louis.bossart@...ux.intel.com>
Cc: Takashi Sakamoto <o-takashi@...amocchi.jp>,
alsa-devel@...a-project.org, netdev@...r.kernel.org,
Richard Cochran <richardcochran@...il.com>,
linux-kernel@...r.kernel.org, Arnd Bergmann <arnd@...aro.org>,
linux-media@...r.kernel.org
Subject: Re: [alsa-devel] [very-RFC 0/8] TSN driver for the kernel
On Mon, Jun 20, 2016 at 01:08:27PM +0200, Pierre-Louis Bossart wrote:
>
> >Presentation time is either set by
> >a) Local sound card performing capture (in which case it will be 'capture
> > time')
> >b) Local media application sending a stream accross the network
> > (time when the sample should be played out remotely)
> >c) Remote media application streaming data *to* host, in which case it will
> > be local presentation time on local soundcard
> >
> >>This value is dominant to the number of events included in an IEC 61883-1
> >>packet. If this TSN subsystem decides it, most of these items don't need
> >>to be in ALSA.
> >
> >Not sure if I understand this correctly.
> >
> >TSN should have a reference to the timing-domain of each *local*
> >sound-device (for local capture or playback) as well as the shared
> >time-reference provided by gPTP.
> >
> >Unless an End-station acts as GrandMaster for the gPTP-domain, time set
> >forth by gPTP is inmutable and cannot be adjusted. It follows that the
> >sample-frequency of the local audio-devices must be adjusted, or the
> >audio-streams to/from said devices must be resampled.
>
> The ALSA API provides support for 'audio' timestamps
> (playback/capture rate defined by audio subsystem) and 'system'
> timestamps (typically linked to TSC/ART) with one option to take
> synchronized timestamps should the hardware support them.
Ok, this sounds promising, and very much in line with what AVB would need.
> The intent was that the 'audio' timestamps are translated to a
> shared time reference managed in userspace by gPTP, which in turn
> would define if (adaptive) audio sample rate conversion is needed.
> There is no support at the moment for a 'play_at' function in ALSA,
> only means to control a feedback loop.
Ok, I understand that the 'play_at' is difficult to obtain, but it sounds
like it is doable to achieve something useful.
Looks like I will be looking into what to put in the .trigger-handler in
the ALSA shim and experimenting with this to see how it make sense to
connect it from the TSN-stream.
Thanks!
--
Henrik Austad
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