[gst-embedded] noise and stuttering
Jan Schmidt
thaytan at noraisin.net
Fri Aug 1 12:08:27 PDT 2008
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: [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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> >> Jan Schmidt <thaytan at noraisin.net>
> >>
> >>
> >>
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Jan Schmidt <thaytan at noraisin.net>
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