[gst-embedded] noise and stuttering
Raj Swaminathan
rajshyam at gmail.com
Mon Aug 4 14:59:58 PDT 2008
Thanks for all your help everyone. I have a faint idea of what i need to do
based off ur hints .... ill work on it and will keep u posted ...
On Sun, Aug 3, 2008 at 8:19 PM, Zhao Liang-E3423C <E3423C at motorola.com>wrote:
> so what's the rootcause of stutter? Is it caused by data drop or playing
> unsmoothly by audio driver?
>
> From your test on OSSink, it seems decoder is slower than playback,
> and gstaudiosink drop data and always output zero data. I think you need
> open some logs (such as ossink) to get accurate information, and find the
> rootcause.
>
> Zhao Liang
> ------------------------------
> *From:* gstreamer-embedded-bounces at lists.sourceforge.net [mailto:
> gstreamer-embedded-bounces at lists.sourceforge.net] *On Behalf Of *Dennis
> Fleming
> *Sent:* Saturday, August 02, 2008 4:19 AM
> *To:* gstreamer-embedded at lists.sourceforge.net
> *Subject:* Re: [gst-embedded] noise and stuttering
>
> It's interesting that I am getting the opposite problem. Ie. stutters
> for wav and not for MP3. It looks like we were optimizing internal buffers
> for 44.1 kHz. However, 22.05kHz had problems with buffer-time=10000 and
> latency-time=100. Going back to the defaults 220xkHz worked but 44.1 was
> sensitive to activity on the system. I'll try the sync fix to see what
> happens on my device.
>
> Dennis
>
> ----- Original Message ----
> From: Jan Schmidt <thaytan at noraisin.net>
> To: Raj Swaminathan <rajshyam at gmail.com>
> Cc: gstreamer-embedded at lists.sourceforge.net
> Sent: Friday, August 1, 2008 12:08:27 PM
> Subject: Re: [gst-embedded] noise and stuttering
>
>
> 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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> > >> > >>
> > >> > >
> > >> >
> > >> >
> > >> >
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> > >> --
> > >> Jan Schmidt <thaytan at noraisin.net>
> > >>
> > >>
> > >>
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> Jan Schmidt <thaytan at noraisin.net>
>
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