noise and stuttering

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Hi Raj,

Have you tried sync=false in the osssink?  I'm not sure that this is
within your purposes but in some cases that removes the glitches.

Daniel.

On Fri, Aug 1, 2008 at 12:41 PM, Raj Swaminathan <rajshyam at gmail.com> wrote:
>
> 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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