Any help on this?<br><br>
<div class="gmail_quote">On Wed, Jul 27, 2011 at 11:25 AM, Vallabha Hampiholi <span dir="ltr"><<a href="http://vallabha.pa">vallabha.pa</a>@<a href="http://googlemail.com">googlemail.com</a>></span> wrote:<br>
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<div>We are now running PA with high resolution timers enabled in the kernel.</div>
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<div>Yet I still can hear the audio glitches.</div>
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<div>Attached are new traces.<br><br></div>
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<div class="gmail_quote">On Sun, Jul 24, 2011 at 7:05 PM, Vallabha Hampiholi <span dir="ltr"><<a href="http://vallabha.pa/" target="_blank">vallabha.pa</a>@<a href="http://googlemail.com/" target="_blank">googlemail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Dear Experts,<br><br>I would like to know a few more things:<br><br>Please pardon me, as I started working PulseAudio recently and havent<br>
gone through the source code yet, before asking these questions due to<br>lack of time.<br><br>1. How is audio data handled between PulseAudio daemon and PulseAudio<br>clients? What is the minimum data that should be transfrered before<br>
PulseAudio starts sending the audio data to ALSA hardware?<br><br>2. Is the above determined by the values of fragment-size and number<br>of fragments in the daemon.conf file?<br><br>3. I see that the default values of the above parameters are 25 msec<br>
and 4. So does this mean that, PulseAudio does not start processing<br>data until 100msec of data is available?<br><br>4. Also from (3) what should be the ideal value for the latency-time<br>for pulsesink? 100msec or 25msec?<br>
<br>Thank you in advance.<br><br>-Rgds<br><font color="#888888">Vallabha<br></font>
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<div><br>On 7/23/11, Vallabha Hampiholi <<a href="http://vallabha.pa/" target="_blank">vallabha.pa</a>@<a href="http://googlemail.com/" target="_blank">googlemail.com</a>> wrote:<br>> Thank you Pierre and Colin.<br>
><br>> @Pierre: Should both parameters be set to -1? Upon gst-inspecting the<br>> pulsesink element, i get following information:<br>> buffer-time : Size of audio buffer in microseconds<br>> flags: readable, writable<br>
> Integer64. Range: 1 - 9223372036854775807 Default:<br>> 200000 Current: 200000<br>> latency-time : Audio latency in microseconds<br>> flags: readable, writable<br>
> Integer64. Range: 1 - 9223372036854775807 Default:<br>> 10000 Current: 10000<br>> Seems like these parameters will not accept negative values.<br>><br>> I am working on an embedded environment, and taking the default vaues of<br>
> latency and buffer time results in buffer underrun.<br>><br>> -Rgds<br>> Vallabha<br>> On Fri, Jul 22, 2011 at 11:43 PM, pl bossart<br>> <<a href="mailto:bossart.nospam@gmail.com" target="_blank">bossart.nospam@gmail.com</a>>wrote:<br>
><br>>> > Can anyone let me know as what criterions should be considered while<br>>> setting<br>>> > the latency-time and buffer-time parameters for the GST element<br>>> pulsesink?<br>>><br>
>> The parameter names are a bit misleading.<br>>> latency-time only deals with the amount of data exchanged between<br>>> pulsesink and pulseaudio. It doesn't really represent the latency.<br>>> buffer-time should be the total buffering/latency you want for your<br>
>> audio chain. If you don't care about it, set it to -1 to reduce the<br>>> number of wakes and decrease power consumption<br>>> -Pierre<br>>> _______________________________________________<br>
>> pulseaudio-discuss mailing list<br>>> <a href="mailto:pulseaudio-discuss@lists.freedesktop.org" target="_blank">pulseaudio-discuss@lists.freedesktop.org</a><br>>> <a href="http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss" target="_blank">http://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss</a><br>
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