Re: [PATCH 17/29] staging: bcm2835-audio: Add 10ms period constraint [Resend in plain text...]

Mike Brady
2018-10-13T15:00:43Z

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 <mikebrady@eircom.net> 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 <mikebrady@eircom.net> wrote:
>>
>> Hi Takashi.
>>
>>> On 9 Oct 2018, at 14:44, Takashi Iwai <tiwai@suse.de> 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
>>
>