<div dir="ltr"><br>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 ...<br><br><br><div class="gmail_quote">On Sun, Aug 3, 2008 at 8:19 PM, Zhao Liang-E3423C <span dir="ltr"><<a href="mailto:E3423C@motorola.com">E3423C@motorola.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div dir="ltr" align="left"><font color="#0000ff" face="Courier New"><span>so what's the rootcause of </span></font><span><font color="#0000ff" face="Courier New">stutter? Is
it caused by data drop or playing unsmoothly by audio
driver?</font></span></div>
<div dir="ltr" align="left"><span></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Courier New">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 color="#0000ff" face="Courier New">get accurate information,
and find the rootcause.</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Courier New"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Courier New">Zhao Liang</font></span></div>
<div dir="ltr" align="left" lang="en-us">
<hr>
<font size="2" face="Tahoma"><div class="Ih2E3d"><b>From:</b>
<a href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net" target="_blank">gstreamer-embedded-bounces@lists.sourceforge.net</a>
[mailto:<a href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net" target="_blank">gstreamer-embedded-bounces@lists.sourceforge.net</a>] <b>On Behalf Of
</b>Dennis Fleming<br></div><b>Sent:</b> Saturday, August 02, 2008 4:19
AM<div class="Ih2E3d"><br><b>To:</b> <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">gstreamer-embedded@lists.sourceforge.net</a><br></div><div><div></div><div class="Wj3C7c"><b>Subject:</b> Re:
[gst-embedded] noise and stuttering<br></div></div></font><br></div><div><div></div><div class="Wj3C7c">
<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 <<a href="mailto:thaytan@noraisin.net" target="_blank">thaytan@noraisin.net</a>><br>To:
Raj Swaminathan <<a href="mailto:rajshyam@gmail.com" target="_blank">rajshyam@gmail.com</a>><br>Cc:
<a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">gstreamer-embedded@lists.sourceforge.net</a><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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">thijsvermeir@gmail.com</a>><br>>
>> > To: Dennis
Fleming <<a href="mailto:arsantiqua@sbcglobal.net" target="_blank">arsantiqua@sbcglobal.net</a>><br>>
>> > Cc: <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">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" target="_blank">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" target="_blank">thijsvermeir@gmail.com</a>><br>>
>> > > To: Zhao
Liang-E3423C <<a href="mailto:E3423C@motorola.com" target="_blank">E3423C@motorola.com</a>><br>>
>> > > Cc: Dennis
Fleming <<a href="mailto:arsantiqua@sbcglobal.net" target="_blank">arsantiqua@sbcglobal.net</a>>;<br>>
>> > > <a href="mailto:gstreamer-embedded@lists.sourceforge.net" target="_blank">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" target="_blank">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" target="_blank">gstreamer-embedded-bounces@lists.sourceforge.net</a><br>>
>> >
>><br>>
[mailto:<a href="mailto:gstreamer-embedded-bounces@lists.sourceforge.net" target="_blank">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" target="_blank">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 <a href="http://2.6.22.19" target="_blank">2.6.22.19</a><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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