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