<div dir="ltr"><br>Can u also explain why the async fix helped ? <br><br><div class="gmail_quote">On Fri, Aug 1, 2008 at 1:23 PM, Raj Swaminathan <span dir="ltr"><<a href="mailto:rajshyam@gmail.com">rajshyam@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr"><br>Hey Dan,<br><br>That worked ... its stuttering .. but keeps playing .... thanks a ton ! <br>
<br>osssink: wav file play fine <br> mp3 files, http links stutter but output sound when buffer-time=1000 latency-time=100 sync=false <br>
<br> <br>Is there any such fix for esdsink ?? <br>Are there more properties that can be modified to stop the stuttering ?? <br><br>regards,<br><font color="#888888">raj</font><div><div></div><div class="Wj3C7c">
<br><br><br><div class="gmail_quote">On Fri, Aug 1, 2008 at 12:52 PM, Daniel Charles <span dir="ltr"><<a href="mailto:dcharlesm@gmail.com" target="_blank">dcharlesm@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Raj,<br>
<br>
Have you tried sync=false in the osssink? I'm not sure that this is<br>
within your purposes but in some cases that removes the glitches.<br>
<font color="#888888"><br>
Daniel.<br>
</font><div><div></div><div><br>
On Fri, Aug 1, 2008 at 12:41 PM, Raj Swaminathan <<a href="mailto:rajshyam@gmail.com" target="_blank">rajshyam@gmail.com</a>> wrote:<br>
><br>
> Hi everyone,<br>
><br>
> Im having stuttering and stopping issues with gstreamer on the OMAP 2430...<br>
> I am using an NFS mounted file system via ethernet ...<br>
><br>
> osssink: WAV files play without an issue.<br>
> mp3 files output sound and stop after a few seconds if i set<br>
> buffer-time=1000 and latency-time=100<br>
> mp3 files do not output sound without the settings above.<br>
> streaming music from http links do not work under any<br>
> setting.<br>
><br>
> esdsink: WAV files do not play.<br>
> mp3 files play nicely.<br>
> streaming music from http links output sound and stop after a<br>
> few seconds.<br>
><br>
> My sources: filesrc, souphttpsrc<br>
> My decoders: wavparse, mad<br>
><br>
> Ive experimented with placing queues before decoding and before sending<br>
> audio to the sink. Trying both ways or either/or, do not impact the output<br>
> signifcantly.<br>
> Can anyone provide some suggestions?<br>
><br>
> Thanks for your help so far.<br>
><br>
> regards,<br>
> raj<br>
><br>
> On Thu, Jul 31, 2008 at 4:42 AM, Jan Schmidt <<a href="mailto:thaytan@noraisin.net" target="_blank">thaytan@noraisin.net</a>> wrote:<br>
>><br>
>> On Wed, 2008-07-30 at 08:50 -0700, Dennis Fleming wrote:<br>
>> > First off: Thank to you and Zhoa-Lang for getting back so quickly.<br>
>> > I'm so busy I forgot my manners.<br>
>> ><br>
>> > Testing to find the parameters I have I used decodebin, but in the<br>
>> > program itself uses playbin with the same effect. The only variation<br>
>> > is that I set the sink property to alsasink since that seems the only<br>
>> > way to set buffer-time and latency-time properties. Also, it seems<br>
>> > counter-intuitive to me that an uncompressed WAV file should have<br>
>> > problems keeping up while MP3s with the same sampling frequency and<br>
>> > word size have none. And yet the artifacts are indicative of dropped<br>
>> > buffers.<br>
>><br>
>> If the bottleneck is retrieving data from the input location, then it's<br>
>> entirely feasible. What's your data store? SD card, NFS? A WAV file<br>
>> might be 10 or more times more data to read and cause read stalls, where<br>
>> the smaller mp3 can be read in fine and decoded in memory with no<br>
>> further problems.<br>
>><br>
>> J.<br>
>><br>
>> > Dennis<br>
>> ><br>
>> ><br>
>> > ----- Original Message ----<br>
>> > From: Thijs Vermeir <<a href="mailto:thijsvermeir@gmail.com" target="_blank">thijsvermeir@gmail.com</a>><br>
>> > To: Dennis Fleming <<a href="mailto:arsantiqua@sbcglobal.net" target="_blank">arsantiqua@sbcglobal.net</a>><br>
>> > Cc: <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">gstreamer-embedded@lists.sourceforge.net</a><br>
>> > Sent: Tuesday, July 29, 2008 3:59:30 PM<br>
>> > Subject: Re: [gst-embedded] noise and stuttering<br>
>> ><br>
>> > Hi,<br>
>> ><br>
>> > On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming<br>
>> > <<a href="mailto:arsantiqua@sbcglobal.net" target="_blank">arsantiqua@sbcglobal.net</a>> wrote:<br>
>> > > The interesting thing is that uncompressed WAV files are causing the<br>
>> > problem<br>
>> > > while MP3s were fixed by setting the buffer-time and latency-time to<br>
>> > values<br>
>> > > smaller than found on a desktop. What would adding a queue do to<br>
>> > latency<br>
>> > > through the system?<br>
>> ><br>
>> > There is no latency in this case because there are no live-sources.<br>
>> > [1]<br>
>> ><br>
>> > > Also, I suppose, that I will need to break up the<br>
>> > > playbin and create a pipeline myself, yes?<br>
>> ><br>
>> > playbin has the queue elements on the correct location, no changes<br>
>> > needed.<br>
>> > You where already using a custom pipeline, no?<br>
>> ><br>
>> > Gr,<br>
>> ><br>
>> > [1]<br>
>> ><br>
>> > <a href="http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup" target="_blank">http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup</a><br>
>> ><br>
>> > ><br>
>> > > Dennis<br>
>> > ><br>
>> > > ----- Original Message ----<br>
>> > > From: Thijs Vermeir <<a href="mailto:thijsvermeir@gmail.com" target="_blank">thijsvermeir@gmail.com</a>><br>
>> > > To: Zhao Liang-E3423C <<a href="mailto:E3423C@motorola.com" target="_blank">E3423C@motorola.com</a>><br>
>> > > Cc: Dennis Fleming <<a href="mailto:arsantiqua@sbcglobal.net" target="_blank">arsantiqua@sbcglobal.net</a>>;<br>
>> > > <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">gstreamer-embedded@lists.sourceforge.net</a><br>
>> > > Sent: Tuesday, July 29, 2008 2:46:42 AM<br>
>> > > Subject: Re: [gst-embedded] noise and stuttering<br>
>> > ><br>
>> > > Hi,<br>
>> > ><br>
>> > > On Tue, Jul 29, 2008 at 11:15 AM, Zhao Liang-E3423C<br>
>> > <<a href="mailto:E3423C@motorola.com" target="_blank">E3423C@motorola.com</a>><br>
>> > > wrote:<br>
>> > >> What's the rootcause of noise and stuttering ?<br>
>> > ><br>
>> > > Now you are using only 1 thread for all the elements and if the<br>
>> > > filesrc or the decoder is too slow sometimes<br>
>> > > you don't have time to catch up. By adding the queue you put the<br>
>> > sink<br>
>> > > in another thread and now the filesrc+decoder can<br>
>> > > do some decoding in advance.<br>
>> > ><br>
>> > > Gr,<br>
>> > > Thijs<br>
>> > ><br>
>> > >><br>
>> > >> For normal playback, it should not have issues. If decoder didn't<br>
>> > drop<br>
>> > >> data, I think alsasink did it.<br>
>> > >> By gstaudiosink mechanism, it will drop data replaced with blank<br>
>> > data when<br>
>> > >> data is late. I guess the rootcause is that.<br>
>> > >><br>
>> > >> If that, I have no ideas except adding a queue before alsasink, and<br>
>> > when<br>
>> > >> queue is empty, pause the pipeline, it will not cause dropout, but<br>
>> > still<br>
>> > >> discontinous.<br>
>> > >><br>
>> > >> Zhao liang<br>
>> > >> ________________________________<br>
>> > >> From: <a href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net" target="_blank">gstreamer-embedded-bounces@lists.sourceforge.net</a><br>
>> > >> [mailto:<a href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net" target="_blank">gstreamer-embedded-bounces@lists.sourceforge.net</a>] On Behalf<br>
>> > Of<br>
>> > >> Dennis Fleming<br>
>> > >> Sent: Tuesday, July 29, 2008 4:37 AM<br>
>> > >> To: <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">gstreamer-embedded@lists.sourceforge.net</a><br>
>> > >> Subject: [gst-embedded] noise and stuttering<br>
>> > >><br>
>> > >> I'm trying to create an audio player on an IMX31 target and I've<br>
>> > found a<br>
>> > >> discrepancy in the output of various formats. If I send MP3 data I<br>
>> > have<br>
>> > >> to<br>
>> > >> set the buffer-time and latency-time to 10000 and 100 respectively<br>
>> > to play<br>
>> > >> without severe dropouts. However WAV files still have drop-out at<br>
>> > a<br>
>> > >> consistent rate (about 1 per 10 sec). Are there some general<br>
>> > features I'm<br>
>> > >> missing or is there some guidance on the buffer-time/latency time<br>
>> > that<br>
>> > >> would<br>
>> > >> account for this difference?<br>
>> > >><br>
>> > >> Linux <a href="http://2.6.22.19" target="_blank">2.6.22.19</a><br>
>> > >> gstreamer 0.10.17 (open-embedded)<br>
>> > >> gst-launch filesrc location=<file> ! decodebin ! alsasink<br>
>> > >> buffer-time=10000<br>
>> > >> latency-time=100<br>
>> > >><br>
>> > >> Dennis<br>
>> > >><br>
>> > >><br>
>> ><br>
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>> --<br>
>> Jan Schmidt <<a href="mailto:thaytan@noraisin.net" target="_blank">thaytan@noraisin.net</a>><br>
>><br>
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</div></div></blockquote></div><br></div></div></div>
</blockquote></div><br></div>