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<DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff><SPAN
class=717470801-04082008>so what's the rootcause of </SPAN></FONT><SPAN
class=717470801-04082008><FONT face="Courier New" color=#0000ff>stutter? Is
it caused by data drop or playing unsmoothly by audio
driver?</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=717470801-04082008></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=717470801-04082008><FONT face="Courier New"
color=#0000ff>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
</FONT> <FONT face="Courier New" color=#0000ff>get accurate information,
and find the rootcause.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=717470801-04082008><FONT face="Courier New"
color=#0000ff></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=717470801-04082008><FONT face="Courier New"
color=#0000ff>Zhao Liang</FONT></SPAN></DIV>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B>
gstreamer-embedded-bounces@lists.sourceforge.net
[mailto:gstreamer-embedded-bounces@lists.sourceforge.net] <B>On Behalf Of
</B>Dennis Fleming<BR><B>Sent:</B> Saturday, August 02, 2008 4:19
AM<BR><B>To:</B> gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> Re:
[gst-embedded] noise and stuttering<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">
<DIV>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.<BR><BR>Dennis<BR></DIV>
<DIV
style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman,new york,times,serif"><BR>
<DIV style="FONT-SIZE: 10pt; FONT-FAMILY: arial,helvetica,sans-serif">-----
Original Message ----<BR>From: Jan Schmidt <thaytan@noraisin.net><BR>To:
Raj Swaminathan <rajshyam@gmail.com><BR>Cc:
gstreamer-embedded@lists.sourceforge.net<BR>Sent: Friday, August 1, 2008
12:08:27 PM<BR>Subject: Re: [gst-embedded] noise and stuttering<BR><BR><BR>On
Fri, 2008-08-01 at 13:51 -0500, Raj Swaminathan wrote:<BR>> <BR>> Can u
also explain why the async fix helped ? <BR><BR>It helps because it deactivates
the clock-synching in the audiosink,<BR>which means that the file plays at the
speed it manages to read, decode<BR>and output the samples -> that's slower
than real-time in this case,<BR>which is why you're getting
'stuttering'.<BR><BR>Measure how long the file takes to play with 'time'
compared to the<BR>duration of the file.<BR><BR>J.<BR><BR>> <BR>> On Fri,
Aug 1, 2008 at 1:23 PM, Raj Swaminathan <<A href="mailto:rajshyam@gmail.com"
ymailto="mailto:rajshyam@gmail.com">rajshyam@gmail.com</A>><BR>>
wrote:<BR>> <BR>>
Hey Dan,<BR>> <BR>>
That worked ... its stuttering .. but keeps playing ....<BR>>
thanks a ton ! <BR>>
<BR>> osssink: wav file play fine <BR>>
mp3 files,
http links stutter but output sound<BR>> when
buffer-time=1000 latency-time=100 sync=false <BR>>
<BR>>
<BR>> Is there any such fix for esdsink ??
<BR>> Are there more properties that can be
modified to stop the<BR>> stuttering ??
<BR>> <BR>>
regards,<BR>> raj<BR>>
<BR>> <BR>>
<BR>> <BR>> On Fri,
Aug 1, 2008 at 12:52 PM, Daniel Charles<BR>>
<<A href="mailto:dcharlesm@gmail.com"
ymailto="mailto:dcharlesm@gmail.com">dcharlesm@gmail.com</A>>
wrote:<BR>> Hi
Raj,<BR>>
<BR>> Have you tried
sync=false in the osssink? I'm not<BR>>
sure that this is<BR>>
within your purposes but in some cases that
removes<BR>> the
glitches.<BR>>
<BR>>
Daniel.<BR>>
<BR>> <BR>>
On Fri, Aug 1, 2008 at 12:41
PM, Raj Swaminathan<BR>>
<<A href="mailto:rajshyam@gmail.com"
ymailto="mailto:rajshyam@gmail.com">rajshyam@gmail.com</A>>
wrote:<BR>>
><BR>> > Hi
everyone,<BR>>
><BR>> > Im
having stuttering and stopping issues with<BR>>
gstreamer on the OMAP 2430...<BR>>
> I am using an NFS mounted file
system via<BR>>
ethernet ...<BR>>
><BR>> >
osssink: WAV files play without an issue.<BR>>
>
mp3 files output sound and stop after<BR>>
a few seconds if i set<BR>>
> buffer-time=1000 and
latency-time=100<BR>>
> mp3 files do not output
sound without<BR>> the
settings above.<BR>>
> streaming music from http
links do not<BR>> work
under any<BR>> >
setting.<BR>>
><BR>> >
esdsink: WAV files do not play.<BR>>
> mp3 files
play nicely.<BR>>
> streaming music from http
links output<BR>>
sound and stop after a<BR>>
> few seconds.<BR>>
><BR>> >
My sources: filesrc, souphttpsrc<BR>>
> My decoders: wavparse, mad<BR>>
><BR>>
> Ive experimented with placing queues before
decoding<BR>> and
before sending<BR>>
> audio to the sink. Trying both ways or either/or, do<BR>>
not impact the output<BR>>
>
signifcantly.<BR>>
> Can anyone provide some suggestions?<BR>>
><BR>>
> Thanks for your help so far.<BR>>
><BR>>
> regards,<BR>>
> raj<BR>>
><BR>>
> On Thu, Jul 31, 2008 at 4:42 AM, Jan
Schmidt<BR>> <<A
href="mailto:thaytan@noraisin.net"
ymailto="mailto:thaytan@noraisin.net">thaytan@noraisin.net</A>>
wrote:<BR>>
>><BR>> >>
On Wed, 2008-07-30 at 08:50 -0700, Dennis Fleming<BR>>
wrote:<BR>>
>> > First off: Thank to you and Zhoa-Lang
for getting<BR>> back
so quickly.<BR>>
>> > I'm so busy I forgot my manners.<BR>>
>> ><BR>>
>> > Testing to find the parameters
I have I used<BR>>
decodebin, but in the<BR>>
>> > program itself uses playbin with the same
effect.<BR>> The only
variation<BR>>
>> > is that I set the sink property to alsasink since<BR>>
that seems the
only<BR>> >>
> way to set buffer-time and latency-time<BR>>
properties. Also, it seems<BR>>
>> > counter-intuitive
to me that an uncompressed WAV<BR>>
file should have<BR>>
>> > problems keeping up while MP3s with the
same<BR>> sampling
frequency and<BR>>
>> > word size have none. And yet the artifacts are<BR>>
indicative of
dropped<BR>> >>
> buffers.<BR>>
>><BR>> >>
If the bottleneck is retrieving data from the input<BR>>
location, then it's<BR>>
>> entirely feasible. What's
your data store? SD card,<BR>>
NFS? A WAV file<BR>>
>> might be 10 or more times more data to read
and<BR>> cause read
stalls, where<BR>>
>> the smaller mp3 can be read in fine and decoded in<BR>>
memory with no<BR>>
>> further
problems.<BR>>
>><BR>> >>
J.<BR>>
>><BR>> >>
> Dennis<BR>>
>> ><BR>>
>> ><BR>>
>> > ----- Original Message ----<BR>>
>> > From: Thijs Vermeir <<A
href="mailto:thijsvermeir@gmail.com"
ymailto="mailto:thijsvermeir@gmail.com">thijsvermeir@gmail.com</A>><BR>>
>> > To: Dennis
Fleming <<A href="mailto:arsantiqua@sbcglobal.net"
ymailto="mailto:arsantiqua@sbcglobal.net">arsantiqua@sbcglobal.net</A>><BR>>
>> > Cc: <A
href="mailto:gstreamer-embedded@lists.sourceforge.net"
ymailto="mailto:gstreamer-embedded@lists.sourceforge.net">gstreamer-embedded@lists.sourceforge.net</A><BR>>
>> > Sent: Tuesday,
July 29, 2008 3:59:30 PM<BR>>
>> > Subject: Re: [gst-embedded] noise and
stuttering<BR>>
>> ><BR>>
>> > Hi,<BR>>
>> ><BR>>
>> > On Tue, Jul 29, 2008 at 5:43 PM, Dennis Fleming<BR>>
>> > <<A
href="mailto:arsantiqua@sbcglobal.net"
ymailto="mailto:arsantiqua@sbcglobal.net">arsantiqua@sbcglobal.net</A>>
wrote:<BR>> >>
> > The interesting thing is that uncompressed WAV<BR>>
files are causing the<BR>>
>> >
problem<BR>> >>
> > while MP3s were fixed by setting the<BR>>
buffer-time and latency-time to<BR>>
>> >
values<BR>> >>
> > smaller than found on a desktop. What would<BR>>
adding a queue do to<BR>>
>> >
latency<BR>> >>
> > through the system?<BR>>
>> ><BR>>
>> > There is no latency in this case because
there<BR>> are no
live-sources.<BR>>
>> > [1]<BR>>
>> ><BR>>
>> > > Also, I suppose, that I will need to break up<BR>>
the<BR>>
>> > > playbin and create a
pipeline myself, yes?<BR>>
>> ><BR>>
>> > playbin has the queue elements on the correct<BR>>
location, no
changes<BR>> >>
> needed.<BR>>
>> > You where already using a custom pipeline, no?<BR>>
>> ><BR>>
>> > Gr,<BR>>
>> ><BR>>
>> > [1]<BR>>
>> ><BR>>
>> ><BR>>
<A
href="http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup"
target=_blank>http://webcvs.freedesktop.org/gstreamer/gstreamer/docs/design/draft-latency.txt?view=markup</A><BR>>
>> ><BR>>
>> >
><BR>> >>
> > Dennis<BR>>
>> > ><BR>>
>> > > ----- Original Message ----<BR>>
>> > > From: Thijs Vermeir
<<A href="mailto:thijsvermeir@gmail.com"
ymailto="mailto:thijsvermeir@gmail.com">thijsvermeir@gmail.com</A>><BR>>
>> > > To: Zhao
Liang-E3423C <<A href="mailto:E3423C@motorola.com"
ymailto="mailto:E3423C@motorola.com">E3423C@motorola.com</A>><BR>>
>> > > Cc: Dennis
Fleming <<A href="mailto:arsantiqua@sbcglobal.net"
ymailto="mailto:arsantiqua@sbcglobal.net">arsantiqua@sbcglobal.net</A>>;<BR>>
>> > > <A
href="mailto:gstreamer-embedded@lists.sourceforge.net"
ymailto="mailto:gstreamer-embedded@lists.sourceforge.net">gstreamer-embedded@lists.sourceforge.net</A><BR>>
>> > > Sent:
Tuesday, July 29, 2008 2:46:42 AM<BR>>
>> > > Subject: Re: [gst-embedded] noise
and<BR>>
stuttering<BR>>
>> > ><BR>>
>> > > Hi,<BR>>
>> > ><BR>>
>> > > On Tue, Jul 29, 2008 at 11:15 AM,
Zhao<BR>>
Liang-E3423C<BR>>
>> > <<A href="mailto:E3423C@motorola.com"
ymailto="mailto:E3423C@motorola.com">E3423C@motorola.com</A>><BR>>
>> > >
wrote:<BR>> >>
> >> What's the rootcause of noise and stuttering ?<BR>>
>> >
><BR>> >>
> > Now you are using only 1 thread for all the<BR>>
elements and if the<BR>>
>> > > filesrc or
the decoder is too slow sometimes<BR>>
>> > > you don't have time to catch up. By
adding the<BR>> queue
you put the<BR>>
>> > sink<BR>>
>> > > in another thread and now the
filesrc+decoder<BR>>
can<BR>> >> >
> do some decoding in advance.<BR>>
>> > ><BR>>
>> > > Gr,<BR>>
>> > > Thijs<BR>>
>> > ><BR>>
>> >
>><BR>> >>
> >> For normal playback, it should not have<BR>>
issues. If decoder
didn't<BR>> >>
> drop<BR>>
>> > >> data, I think alsasink did it.<BR>>
>> > >> By gstaudiosink
mechanism, it will drop data<BR>>
replaced with blank<BR>>
>> > data when<BR>>
>> > >> data is late. I guess the
rootcause is that.<BR>>
>> > >><BR>>
>> > >> If that, I have no ideas except adding a
queue<BR>> before
alsasink, and<BR>>
>> > when<BR>>
>> > >> queue is empty, pause the pipeline, it
will<BR>> not cause
dropout, but<BR>>
>> > still<BR>>
>> > >> discontinous.<BR>>
>> > >><BR>>
>> > >> Zhao
liang<BR>> >>
> >> ________________________________<BR>>
>> > >> From:<BR>>
<A
href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net"
ymailto="mailto:gstreamer-embedded-bounces@lists.sourceforge.net">gstreamer-embedded-bounces@lists.sourceforge.net</A><BR>>
>> >
>><BR>>
[mailto:<A href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net"
ymailto="mailto:gstreamer-embedded-bounces@lists.sourceforge.net">gstreamer-embedded-bounces@lists.sourceforge.net</A>]
On Behalf<BR>>
>> > Of<BR>>
>> > >> Dennis Fleming<BR>>
>> > >> Sent: Tuesday, July 29, 2008 4:37
AM<BR>> >> >
>> To: <A href="mailto:gstreamer-embedded@lists.sourceforge.net"
ymailto="mailto:gstreamer-embedded@lists.sourceforge.net">gstreamer-embedded@lists.sourceforge.net</A><BR>>
>> > >> Subject:
[gst-embedded] noise and stuttering<BR>>
>> > >><BR>>
>> > >> I'm trying to create an audio
player on an<BR>>
IMX31 target and I've<BR>>
>> > found a<BR>>
>> > >> discrepancy in the output of various
formats.<BR>>
If I send MP3 data I<BR>>
>> > have<BR>>
>> > >> to<BR>>
>> > >> set the buffer-time and
latency-time to 10000<BR>>
and 100 respectively<BR>>
>> > to play<BR>>
>> > >> without severe dropouts.
However WAV files<BR>>
still have drop-out at<BR>>
>> > a<BR>>
>> > >> consistent rate (about 1 per 10 sec).
Are<BR>> there some
general<BR>> >>
> features I'm<BR>>
>> > >> missing or is there some guidance on the<BR>>
buffer-time/latency
time<BR>> >>
> that<BR>>
>> > >> would<BR>>
>> > >> account for this difference?<BR>>
>> >
>><BR>> >>
> >> Linux 2.6.22.19<BR>>
>> > >> gstreamer 0.10.17
(open-embedded)<BR>>
>> > >> gst-launch filesrc location=<file> !<BR>>
decodebin !
alsasink<BR>> >>
> >> buffer-time=10000<BR>>
>> > >> latency-time=100<BR>>
>> > >><BR>>
>> > >>
Dennis<BR>> >>
> >><BR>>
>> > >><BR>>
>> ><BR>>
>> ><BR>>
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ymailto="mailto:thaytan@noraisin.net">thaytan@noraisin.net</A>><BR>>
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