PulseSink Properties

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Any help on this?

On Wed, Jul 27, 2011 at 11:25 AM, Vallabha Hampiholi <vallabha.pa@
googlemail.com> wrote:

> We are now running PA with high resolution timers enabled in the kernel.
>
> Yet I still can hear the audio glitches.
>
> Attached are new traces.
>
>   On Sun, Jul 24, 2011 at 7:05 PM, Vallabha Hampiholi <vallabha.pa@
> googlemail.com> wrote:
>
>> 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
>> >> _______________________________________________
>> >> pulseaudio-discuss mailing list
>> >> pulseaudio-discuss at lists.freedesktop.org
>> >> http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss
>> >>
>> >
>>
>
>
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