noise and stuttering

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On Fri, 2008-08-01 at 13:51 -0500, Raj Swaminathan wrote:
> 
> Can u also explain why the async fix helped ? 

?It helps because it deactivates the clock-synching in the audiosink,
which means that the file plays at the speed it manages to read, decode
and output the samples -> that's slower than real-time in this case,
which is why you're getting 'stuttering'.

Measure how long the file takes to play with 'time' compared to the
duration of the file.

J.

> 
> 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: 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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>         
> 
> 
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Jan Schmidt <thaytan at noraisin.net>





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