Date:   Thu, 11 Oct 2018 13:53:47 +0100
From:   Mike Brady <mikebrady@...com.net>
To:     Takashi Iwai <tiwai@...e.de>
Cc:     Stefan Wahren <stefan.wahren@...e.com>,
        Greg Kroah-Hartman <gregkh@...uxfoundation.org>,
        Eric Anholt <eric@...olt.net>,
        linux-rpi-kernel@...ts.infradead.org, linux-kernel@...r.kernel.org,
        Phil Elwell <phil@...pberrypi.org>
Subject: Re: [PATCH 17/29] staging: bcm2835-audio: Add 10ms period constraint
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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@...com.net> wrote:
> 
> Hi Takashi.
> 
>> On 9 Oct 2018, at 14:44, Takashi Iwai <tiwai@...e.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
> 

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