Re: [very-RFC 0/8] TSN driver for the kernel

[John Fastabend <john.fastabend@gmail.com>]

On 16-06-13 04:47 AM, Richard Cochran wrote:
> Henrik,
> 
> On Sun, Jun 12, 2016 at 01:01:28AM +0200, Henrik Austad wrote:
>> There are at least one AVB-driver (the AV-part of TSN) in the kernel
>> already,
> 
> Which driver is that?
> 
>> however this driver aims to solve a wider scope as TSN can do
>> much more than just audio. A very basic ALSA-driver is added to the end
>> that allows you to play music between 2 machines using aplay in one end
>> and arecord | aplay on the other (some fiddling required) We have plans
>> for doing the same for v4l2 eventually (but there are other fishes to
>> fry first). The same goes for a TSN_SOCK type approach as well.
> 
> Please, no new socket type for this.
>  
>> What remains
>> - tie to (g)PTP properly, currently using ktime_get() for presentation
>>   time
>> - get time from shim into TSN and vice versa
> 
> ... and a whole lot more, see below.
> 
>> - let shim create/manage buffer
> 
> (BTW, shim is a terrible name for that.)
> 
> [sigh]
> 
> People have been asking me about TSN and Linux, and we've made some
> thoughts about it.  The interest is there, and so I am glad to see
> discussion on this topic.
> 
> Having said that, your series does not even begin to address the real
> issues.  I did not review the patches too carefully (because the
> important stuff is missing), but surely configfs is the wrong
> interface for this.  In the end, we will be able to support TSN using
> the existing networking and audio interfaces, adding appropriate
> extensions.
> 
> Your patch features a buffer shared by networking and audio.  This
> isn't strictly necessary for TSN, and it may be harmful.  The
> Listeners are supposed to calculate the delay from frame reception to
> the DA conversion.  They can easily include the time needed for a user
> space program to parse the frames, copy (and combine/convert) the
> data, and re-start the audio transfer.  A flexible TSN implementation
> will leave all of the format and encoding task to the userland.  After
> all, TSN will some include more that just AV data, as you know.
> 
> Lets take a look at the big picture.  One aspect of TSN is already
> fully supported, namely the gPTP.  Using the linuxptp user stack and a
> modern kernel, you have a complete 802.1AS-2011 solution.
> 
> Here is what is missing to support audio TSN:
> 
> * User Space
> 
> 1. A proper userland stack for AVDECC, MAAP, FQTSS, and so on.  The
>    OpenAVB project does not offer much beyond simple examples.
> 
> 2. A user space audio application that puts it all together, making
>    use of the services in #1, the linuxptp gPTP service, the ALSA
>    services, and the network connections.  This program will have all
>    the knowledge about packet formats, AV encodings, and the local HW
>    capabilities.  This program cannot yet be written, as we still need
>    some kernel work in the audio and networking subsystems.
> 
> * Kernel Space
> 
> 1. Providing frames with a future transmit time.  For normal sockets,
>    this can be in the CMESG data.  For mmap'ed buffers, we will need a
>    new format.  (I think Arnd is working on a new layout.)
> 
> 2. Time based qdisc for transmitted frames.  For MACs that support
>    this (like the i210), we only have to place the frame into the
>    correct queue.  For normal HW, we want to be able to reserve a time
>    window in which non-TSN frames are blocked.  This is some work, but
>    in the end it should be a generic solution that not only works
>    "perfectly" with TSN HW but also provides best effort service using
>    any NIC.
> 

When I looked at this awhile ago I convinced myself that it could fit
fairly well into the DCB stack (DCB is also part of 802.1Q). A lot of
the traffic class to queue mappings and priories could be handled here.
It might be worth taking a look at ./net/sched/mqprio.c and ./net/dcb/.

Unfortunately I didn't get too far along but we probably don't want
another mechanism to map hw queues/tcs/etc if the existing interfaces
work or can be extended to support this.

> 3. ALSA support for tunable AD/DA clocks.  The rate of the Listener's
>    DA clock must match that of the Talker and the other Listeners.
>    Either you adjust it in HW using a VCO or similar, or you do
>    adaptive sample rate conversion in the application. (And that is
>    another reason for *not* having a shared kernel buffer.)  For the
>    Talker, either you adjust the AD clock to match the PTP time, or
>    you measure the frequency offset.
> 
> 4. ALSA support for time triggered playback.  The patch series
>    completely ignore the critical issue of media clock recovery.  The
>    Listener must buffer the stream in order to play it exactly at a
>    specified time.  It cannot simply send the stream ASAP to the audio
>    HW, because some other Listener might need longer.  AFAICT, there
>    is nothing in ALSA that allows you to say, sample X should be
>    played at time Y.
> 
> These are some ideas about implementing TSN.  Maybe some of it is
> wrong (especially about ALSA), but we definitely need a proper design
> to get the kernel parts right.  There is plenty of work to do, but we
> really don't need some hacky, in-kernel buffer with hard coded audio
> formats.
> 
> Thanks,
> Richard
> 

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