On Mon, Jun 13, 2016 at 09:32:10PM +0200, Richard Cochran wrote:
On Mon, Jun 13, 2016 at 03:00:59PM +0200, Henrik Austad wrote:
On Mon, Jun 13, 2016 at 01:47:13PM +0200, Richard Cochran wrote:
Which driver is that?
drivers/net/ethernet/renesas/
That driver is merely a PTP capable MAC driver, nothing more. Although AVB is in the device name, the driver doesn't implement anything beyond the PTP bits.
Yes, I think they do the rest from userspace, not sure though :)
What is the rationale for no new sockets? To avoid cluttering? or do sockets have a drawback I'm not aware of?
The current raw sockets will work just fine. Again, there should be a application that sits in between with the network socket and the audio interface.
So loop data from kernel -> userspace -> kernelspace and finally back to userspace and the media application? I agree that you need a way to pipe the incoming data directly from the network to userspace for those TSN users that can handle it. But again, for media-applications that don't know (or care) about AVB, it should be fed to ALSA/v4l2 directly and not jump between kernel and userspace an extra round.
I get the point of not including every single audio/video encoder in the kernel, but raw audio should be piped directly to alsa. V4L2 has a way of piping encoded video through the system and to the media application (in order to support cameras that to encoding). The same approach should be doable for AVB, no? (someone from alsa/v4l2 should probably comment on this)
Why is configfs wrong?
Because the application will use the already existing network and audio interfaces to configure the system.
Configuring this via the audio-interface is going to be a challenge since you need to configure the stream through the network before you can create the audio interface. If not, you will have to either drop data or block the caller until the link has been fully configured.
This is actually the reason why configfs is used in the series now, as it allows userspace to figure out all the different attributes and configure the link before letting ALSA start pushing data.
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.
Yes, I thought so, which is also why I have put that to the side and why I'm using ktime_get() for timestamps at the moment. There's also the issue of hooking the time into ALSA/V4L2
So lets get that issue solved before anything else. It is absolutely essential for TSN. Without the synchronization, you are only playing audio over the network. We already have software for that.
Yes, I agree, presentation-time and local time needs to be handled properly. The same for adjusting sample-rate etc. This is a lot of work, so I hope you can understand why I started out with a simple approach to spark a discussion before moving on to the larger bits.
- 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.
Why?
Because user space is right place to place the knowledge of the myriad formats and options.
Se response above, better to let anything but uncompressed raw data trickle through.
the whole point should be to make it as easy for userspace as possible. If you need to tailor each individual media-appliation to use AVB, it is not going to be very useful outside pro-Audio. Sure, there will be challenges, but one key element here should be to *not* require upgrading every single media application.
Then, back to the suggestion of adding a TSN_SOCKET (which you didn't like, but can we agree on a term "raw interface to TSN", and mode of transport can be defined later? ), was to let those applications that are TSN-aware to do what they need to do, whether it is controlling robots or media streams.
First you say you don't want ot upgrade media applications, but then you invent a new socket type. That is a contradiction in terms.
Hehe, no, bad phrasing on my part. I want *both* (hence the shim-interface) :)
Audio apps already use networking, and they already use the audio subsystem. We need to help them get their job done by providing the missing kernel interfaces. They don't need extra magic buffering the kernel. They already can buffer audio data by themselves.
Yes, I know some audio apps "use networking", I can stream netradio, I can use jack to connect devices using RTP and probably a whole lot of other applications do similar things. However, AVB is more about using the network as a virtual sound-card. For the media application, it should not have to care if the device it is using is a soudncard inside the box or a set of AVB-capable speakers somewhere on the network.
- Kernel Space
- 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.)
Ah, I was unaware of this, both CMESG and mmap buffers.
What is the accuracy of deferred transmit? If you have a class A stream, you push out a new frame every 125 us, you may end up with accuracy-constraints lower than that if you want to be able to state "send frame X at time Y".
I have no idea what you are asking here.
I assumed that when you had a mmap'd buffer you'd have to specify a point in time at which the frame should be sent. And since a class A has a 125us interval of sending frames, you have to be able to send frames with enough accuray to that. That's a pretty strict deadline coming from userspace. But as they say, never assume.
I have a lot to dig into, and I've gotten a lot of very useful pointers. I should be busy for a while