[pulseaudio-discuss] PulseSink Properties

[Vallabha Hampiholi]

Dear Experts,

I would like to know a few more things:

Please pardon me, as I started working PulseAudio recently and havent
gone through the source code yet, before asking these questions due to
lack of time.

1. How is audio data handled between PulseAudio daemon and PulseAudio
clients? What is the minimum data that should be transfrered before
PulseAudio starts sending the audio data to ALSA hardware?

2. Is the above determined by the values of fragment-size and number
of fragments in the daemon.conf file?

3. I see that the default values of the above parameters are 25 msec
and 4. So does this mean that, PulseAudio does not start processing
data until 100msec of data is available?

4. Also from (3) what should be the ideal value for the latency-time
for pulsesink? 100msec or 25msec?

Thank you in advance.

-Rgds
Vallabha

On 7/23/11, Vallabha Hampiholi <vallabha.pa at googlemail.com> wrote:
> Thank you Pierre and Colin.
>
> @Pierre: Should both parameters be set to -1? Upon gst-inspecting the
> pulsesink element, i get following information:
> buffer-time         : Size of audio buffer in microseconds
>                         flags: readable, writable
>                         Integer64. Range: 1 - 9223372036854775807 Default:
> 200000 Current: 200000
> latency-time      : Audio latency in microseconds
>                         flags: readable, writable
>                         Integer64. Range: 1 - 9223372036854775807 Default:
> 10000 Current: 10000
> Seems like these parameters will not accept negative values.
>
> I am working on an embedded environment, and taking the default vaues of
> latency and buffer time results in buffer underrun.
>
> -Rgds
> Vallabha
> On Fri, Jul 22, 2011 at 11:43 PM, pl bossart
> <bossart.nospam at gmail.com>wrote:
>
>> > Can anyone let me know as what criterions should be considered while
>> setting
>> > the latency-time and buffer-time parameters for the GST element
>> pulsesink?
>>
>> The parameter names are a bit misleading.
>> latency-time only deals with the amount of data exchanged between
>> pulsesink and pulseaudio. It doesn't really represent the latency.
>> buffer-time should be the total buffering/latency you want for your
>> audio chain. If you don't care about it, set it to -1 to reduce the
>> number of wakes and decrease power consumption
>> -Pierre
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>>
>