Em 07-09-2011 00:37, Devin Heitmueller escreveu:
> On Tue, Sep 6, 2011 at 11:29 PM, Mauro Carvalho Chehab
> <mche...@redhat.com> wrote:
>>> Basically the above starts at the *maximum* capture resolution and
>>> works its way down.  One might argue that this heuristic makes more
>>> sense anyway - why *wouldn't* you want the highest quality audio
>>> possible by default (rather than the lowest)?
>>
>> That change makes sense to me. Yet, you should try to disable pulseaudio
>> and see what's the _real_ speed that the audio announces. On Fedora,
>> just removing pulsaudio-oss-plugin (or something like that) is enough.
>>
>> It seems doubtful that my em2820 WinTV USB2 is different than yours.
>> I suspect that pulseaudio is passing the "extra" range, offering to
>> interpolate the data.
> 
> I disabled pulseaudio and the capture device is advertising the exact
> same range (8-48 KHz).  Seems to be behaving the same way as well.


Hmm... just checked with WinTV USB2:
[    3.819236] msp3400 15-0044: MSP3445G-B8 found @ 0x88 (em28xx #0)

While this device uses snd-usb-audio, it is a non-AC97 adapter. So,
we may expect it to be different from yours. 

Its A/D works at a fixed rate of 32 kHZ, according with MSP34x5G datasheet,
so snd-usb-audio is working properly here.

> So while I'm usually willing to blame things on Pulse, this doesn't
> look like the case here.

That's good. One less problem to deal with.

>>> Even with that patch though, I hit severe underrun/overrun conditions
>>> at 30ms of latency (to the point where the audio is interrupted dozens
>>> of times per second).
>>
>> Yes, it is the same here: 30ms on my notebook is not enough for WinTV
>> USB2 (it is OK with HVR-950).
>>
>>> Turned it up to 50ms and it's much better.
>>> That said, of course such a change would impact lipsync, so perhaps we
>>> need to be adjusting the periods instead.
>>
>> We've added a parameter for that on xawtv3 (--alsa-latency). We've 
>> parametrized
>> it at the alsa stream function call. So, all it needs is to add a new 
>> parameter
>> at tvtime config file.
> 
> Ugh.  We really need some sort of heuristic to do this.  It's
> unreasonable to expect users to know about some magic parameter buried
> in a config file which causes it to start working.

Agreed, but still it makes sense to allow users to adjust, as extra delays
might be needed on really slow machines.

> Perhaps a counter
> that increments whenever an underrun is hit, and after a certain
> number it automatically restarts the stream with a higher latency.  Or
> perhaps we're just making some poor choice in terms of the
> buffers/periods for a given rate.
> 
> Devin
> 

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