Hi Takashi. My apologies — t turns out I was wrong. My measurements were systematically wrong due to integer truncation going from 64 bit to 32 bit representation.
Apologies Mike > On 11 Oct 2018, at 13:53, Mike Brady <[email protected]> wrote: > > 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 >> >
