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Re: Suggestion for minimizing audio glitches

From: Alistair MacDonald <al@signedon.com>
Date: Tue, 17 Apr 2012 12:05:47 -0400
Message-ID: <CAJX8r2kxsyw85vZY-AiOM6i-6A6G6f0GR8nrmTLkAw7YDTsz9g@mail.gmail.com>
To: Randell Jesup <randell-ietf@jesup.org>
Cc: public-audio@w3.org
Randell's 1-5 suggestions are very interesting.

I would think putting this behavior on the destination node might be odd.
But I wonder if adding this kind of behavior to a something like the gain
node might be useful?

For example: if I wanted to combine Video-Chat with a DAW (UC-1 & UC3),
then the following issues would be in play...

1) If a VOIP stream stops suddenly, the user might think there was a
pop/click in their audio track. Adding a tail/decay would be a solution.
(Randell's Option 3)
2) Being a DAW, we would need as much CPU as possible. So avoiding the tail
calculation in JavaScript would be ideal.








On Mon, Apr 16, 2012 at 5:31 PM, Randell Jesup <randell-ietf@jesup.org>
wrote:
> On 4/16/2012 4:10 PM, Chris Rogers wrote:
>
>
>
> On Sun, Apr 15, 2012 at 12:22 PM, <lemeslep@free.fr> wrote:
>>
>> On the current Web Audio draft, it is mentionned in 15.2 that "Audio
>> glitches are caused by an interruption of the normal continuous audio
>> stream, resulting in loud clicks and pops. It is considered to be a
>> catastrophic failure of a multi-media system and must be avoided."
>> And I can't agree more with this!
>> I'm currently facing those ugly audio glitches in my project. I'm using
>> Mozilla's Audio Data API at the moment, and I think I know how browsers
>> could help me to mitigate this problem.
>>
>> The clicks and pops are happening because if the audio buffer is underrun
>> by the javascript app, the audio card is not feeded anymore, and so the
card
>> output goes straight from the value of the last sample played to 0.
>> What would be needed is, perhaps as an option in the Javascript audio
node
>> (?), to have the browser automatically feed the audio card by sustaining
the
>> last sample the javascript application sent, when the audio buffer is
>> underrun.
>>
>> That would really go a long way towards minimizing this critical issue.
>
>
> Hi Philippe, I don't think this will help with the glitches.  Using this
> approach, an under-run will still be quite audible.  And it's not a good
> idea to send a constant (non-zero) value out to the audio hardware since
> this represents a "DC offset" and can cause even worse problems.
>
>
> Since underruns may happen no matter what you do (especially if
main-thread
> JS is involved), it's best to minimize the impact of them.  On an
underrun,
> the primary options are:
>
> 1) send 0's (which generally is the audio device default if you don't feed
> it) - clicks/pops
> 2) repeat last sample - classic lost-packet basic VoIP technique; works ok
> in most cases; requires blending at start/end to avoid click/pop.  Often
> done at a reduced volume which makes it less noticable.
> 3) decay - take last sample and decay it to silence to avoid click/pop -
> more useful if you expect continued lack of source.  Can be variant of #2
> where you progressively decay each missing frame.
> 4) fancier VoIP-style packet loss concealment - better than #2; may tend
to
> be voice-centric
> 5) fancier loss concealment using non-voice centric prediction (waving
hands
> here; I'm sure such things exist for good CD/DVD/etc players).
>
> --
> Randell Jesup
> randell-ietf@jesup.org



-- 
Alistair MacDonald
SignedOn, Inc - W3C Audio WG
Boston, MA, (707) 701-3730
al@signedon.com - http://signedon.com
Received on Tuesday, 17 April 2012 16:06:19 GMT

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