Hi Takashi. Just testing out the updated bcm2835 audio driver — it seems that it will underflow at somewhere above about 4410 and below 5120 frames, whereas the present driver is happy down to at least 2000 frames — I haven’t tried lower than about 1700.
Is this change meant to happen? Regards Mike > On 9 Oct 2018, at 16:28, Mike Brady <[email protected]> wrote: > > Hi Takashi. > >> On 9 Oct 2018, at 14:44, Takashi Iwai <[email protected]> wrote: >> >> On Tue, 09 Oct 2018 15:18:15 +0200, >> Mike Brady wrote: >>> >>>>> @Mike: Do you want to write a patch series which upstream "interpolate >>>>> audio delay" and addresses Takashi's comments? >>>>> >>>>> I would help you, in case you have questions about setup a Raspberry Pi >>>>> with Mainline kernel or patch submission. >>>> >>>> Well, the question is who really wants this. The value given by that >>>> patch is nothing but some estimation and might be even incorrect. >>>> >>>> PulseAudio won't need it any longer when you set the BATCH flag. >>>> Then it'll switch from tsched mode to the old mode, and the delay >>>> value would be almost irrelevant. >>> >>> Well, two answers. First, Shairport Sync >>> (https://github.com/mikebrady/shairport-sync) needs it — whenever a >>> packet of audio frames is about to be added to the output queue (at >>> approximately 7.9 millisecond intervals), the delay is checked to >>> try to maintain sync to within a few milliseconds. The BCM2835 audio >>> device is the only one I have yet come across with so much >>> jitter. Whatever other drivers do, the delay they report doesn’t >>> suffer from anything like this level of jitter. >> >> OK, if there is another application using that delay value, it's worth >> to consider providing a fine-grained value. >> >>> The second answer is that the veracity of the ALSA documentation >>> depends on it — any application using the ALSA system for >>> synchronisation will rely on this being an accurate reflection of >>> the situation. AFAIK there is really no workaround it if the >>> application is confined to “safe” ALSA >>> (http://0pointer.de/blog/projects/guide-to-sound-apis). >> >>> On LMKL.org, Takashi wrote: >>> >>>> Date Wed, 19 Sep 2018 11:52:33 +0200 >>>> From Takashi Iwai <> >>>> Subject Re: [PATCH 17/29] staging: bcm2835-audio: Add 10ms period >>>> constraint >>> >>>> [snip] >>> >>>> That's OK, as long as the computation is accurate enough (at least not >>>> exceed the actual position) and is light-weight. >>> >>>> [snip] >>> >>> The overhead is small -- an extra ktime_get() every time a GPU message >>> is sent -- and another call and a few calculations whenever the delay >>> is sought from userland. >>> >>> At 48,000 frames per second, i.e. approximately 20 microseconds per >>> frame, it would take a clock inaccuracy of roughly >>> 20 microseconds in 10 milliseconds -- 2,000 parts per million — to >>> result in an inaccurate estimate. >>> Crystal or resonator-based clocks typically have an inaccuracy of >>> 10s to 100s of parts per million. >>> >>> Finally, to see the effect of the absence and presence of this >>> interpolation, please have a look at this: >>> https://github.com/raspberrypi/firmware/issues/1026#issuecomment-415746016, >>> where a downstream version of this fix was being discussed. >> >> I'm not opposing to the usage of delay value. The attribute is >> provided exactly for such a purpose. It's a good thing (tm). >> >> The potential problem is, however, rather the implementation: it's >> using a system timer for interpolation, which is known to drift from >> the actual clocks. Though, one may say that in such a use case, we >> may ignore the drift since the interpolation is so narrow. > > Yes, that was my thought. I guess another thing in its favour is that this > audio device will always > be in partnership with a processor as part of an SoC, so it will always be > likely to have a reasonably > accurate clock. > >> But another question is whether it should be implemented in each >> driver level. The time-stamping is basically a PCM core >> functionality, and nothing specific to the hardware, especially when >> it's referring to the system timer. > > That’s a fair point. I don’t know what is done in other drivers, but can only > report that with one possible exception, > the DACs used with Shairport Sync by many end users report well-behaved delay > figures, certainly to within two microseconds. I’m afraid I don’t know how > they do it. > >> e.g. you can think in a different way, too: we may put a timestamp at >> each hwptr update, and pass it as-is, instead of updating the >> timestamp at each position query. This will effectively gives the >> accurate position-timestamp pair, and user-space may interpolate as it >> likes, too. > > That’s not a bad idea, and I might take it up on the alsa-devel mailing list, > as you suggest. > >> In anyway, if *this* kind of feature needs to be merged, it's >> definitely to be discussed with the upstream. So, if you're going to >> merge that sort of path, please keep Cc to alsa-devel ML. > > In the meantime, would you think that the balance of convenience lies with > this interpolation scheme? (Finally, I have a patch ready….) > Regards > Mike > >> >> thanks, >> >> Takashi >
