[gst-embedded] noise and stuttering

Raj Swaminathan rajshyam at gmail.com
Fri Aug 1 11:51:54 PDT 2008 

Can u also explain why the async fix helped ?

On Fri, Aug 1, 2008 at 1:23 PM, Raj Swaminathan <rajshyam at gmail.com> wrote:

>
> Hey Dan,
>
> That worked ... its stuttering .. but keeps playing  .... thanks a ton !
>
> osssink: wav file play fine
>              mp3 files, http links stutter but output sound when
> buffer-time=1000 latency-time=100 sync=false
>
>
> Is there any such fix for esdsink ??
> Are there more properties that can be modified to stop the stuttering ??
>
> regards,
> raj
>
>
>
> On Fri, Aug 1, 2008 at 12:52 PM, Daniel Charles <dcharlesm at gmail.com>wrote:
>
>> 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: [gst-embedded] 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: [gst-embedded] 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: [gst-embedded] 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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>> >> > >>
>> >> > >
>> >> >
>> >> >
>> >> >
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>> >> > _______________________________________________ Gstreamer-embedded
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>> >> --
>> >> Jan Schmidt <thaytan at noraisin.net>
>> >>
>> >>
>> >>
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