noise and stuttering

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Removing the clock sync is not an elegant way of fixing this problem,
as Jan said, once you get a faster system you will find playing very
fast issues.  I've investigated this particular problem in a OMAP 3430
some time ago and, under my particular purposes, setting the timestamp
did help to avoid the glitches and the stutter.

I must say that this could not fix your problem in the long term, but
you could try by creating your own audiosink/gstaudioclock depending
on your development.

Daniel.

On Fri, Aug 1, 2008 at 2:08 PM, Jan Schmidt <thaytan at noraisin.net> wrote:
>
> 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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> --
> Jan Schmidt <thaytan at noraisin.net>
>
>


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