noise and stuttering

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