On Sun, Jun 12, 2016 at 01:30:24PM +0900, Takashi Sakamoto wrote:
On Jun 12 2016 12:38, Takashi Sakamoto wrote:
In your patcset, there's no actual codes about how to handle any interrupt contexts (software / hardware), how to handle packet payload, and so on. Especially, for recent sound subsystem, the timing of generating interrupts and which context does what works are important to reduce playback/capture latency and power consumption.
Of source, your intention of this patchset is to show your early concept of TSN feature. Nevertheless, both of explaination and codes are important to the other developers who have little knowledges about TSN, AVB and AES-64 such as me.
Oops. Your 5th patch was skipped by alsa-project.org. I guess that size of the patch is too large to the list service. I can see it: http://marc.info/?l=linux-netdev&m=146568672728661&w=2
As long as seeing the patch, packets are queueing in hrtimer callbacks every 1 seconds.
Actually, the hrtimer fires every 1ms, and that part is something I have to do something about, also because it sends of the same number of frames every time, regardless of how accurate the internal timer is to the rest of the network (there's no backpressure from the networking layer).
(This is a high level discussion and it's OK to ignore it for the moment. When writing packet-oriented drivers for sound subsystem, you need to pay special attention to accuracy of the number of PCM frames transferred currently, and granularity of the number of PCM frames transferred by one operation. In this case, snd_avb_hw, snd_avb_pcm_pointer(), tsn_buffer_write_net() and tsn_buffer_read_net() are involved in this discussion. You can see ALSA developers' struggle in USB audio device class drivers and (of cource) IEC 61883-1/6 drivers.)
Ah, good point. Any particular parts of the USB-subsystem I should start looking at? Knowing where to start looking is a tremendous help
Thanks for the feedback!