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<DIV><SPAN class=032254108-18062008><FONT face="Book Antiqua" color=#0000ff>Why
your source element has a break? do you use live source?</FONT></SPAN></DIV>
<DIV><SPAN class=032254108-18062008><FONT face="Book Antiqua"
color=#0000ff></FONT></SPAN> </DIV>
<DIV><SPAN class=032254108-18062008><FONT face="Book Antiqua" color=#0000ff>I
suggest you pause the pipleline at the moment.</FONT></SPAN></DIV>
<DIV><SPAN class=032254108-18062008><FONT face="Book Antiqua"
color=#0000ff></FONT></SPAN> </DIV>
<DIV><SPAN class=032254108-18062008><FONT face="Book Antiqua"
color=#0000ff>Brad</FONT></SPAN></DIV><BR>
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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<FONT face=Tahoma><B>From:</B> Shenhong Wang [mailto:qch1688@hotmail.com]
<BR><B>Sent:</B> Wednesday, June 18, 2008 4:39 PM<BR><B>To:</B> Zhao Bin-E6223C;
gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> RE: [gst-embedded]
Question on gst_plugin alsasink<BR></FONT><BR></DIV>
<DIV></DIV><BR>Brad,<BR>thanks! <BR>At the moment I didn't put any timestamp on
the frames(buffer) yet.<BR>Generally, the aacdec &alsasink will not play out
any audio frames(packets) after its source element has a break to send audio
frames (packets) to them. It looks the alsasink drop all frames(packets) from
the break. Why? How to fix it? Thanks!<BR> <BR>Best Regards!<BR>Shenhong
WANG<BR><BR><BR><BR>
<BLOCKQUOTE>
<HR id=EC_stopSpelling>
Subject: RE: [gst-embedded] Question on gst_plugin alsasink<BR>Date: Wed, 18
Jun 2008 15:01:06 +0800<BR>From: binzhao@motorola.com<BR>To:
qch1688@hotmail.com; gstreamer-embedded@lists.sourceforge.net<BR><BR>
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<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008>Hi, your issue seems that audio frame is delayed
when arrivering alsasink.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008>basesink will drop the delayed buffer and you
could'nt hear any sound.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008>Please check your timestamp of
buffer.</SPAN></FONT></DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008></SPAN></FONT> </DIV>
<DIV dir=ltr align=left><FONT face=Arial color=#0000ff><SPAN
class=EC_799365806-18062008>Brad</SPAN></FONT></DIV>
<DIV> </DIV><BR>
<DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR>
<FONT face=Tahoma><B>From:</B>
gstreamer-embedded-bounces@lists.sourceforge.net
[mailto:gstreamer-embedded-bounces@lists.sourceforge.net] <B>On Behalf Of
</B>Shenhong Wang<BR><B>Sent:</B> Wednesday, June 18, 2008 2:21
PM<BR><B>To:</B> gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B>
[gst-embedded] Question on gst_plugin alsasink<BR></FONT><BR></DIV>
<DIV></DIV><BR>Dear all,<BR>Now we are using alsasink to play audio on Marvell
PXA310 board. The audio is aac format. The audio frames(packets)
are frequently sent to the aac decoder & alsasink to play out.
Unfortunately only the begining frames can be played out and then nothing is
played out. <BR>If we save those audio frames into a file, the aac
decoder&alsasink can be successfully played out. It means the audio frames
are ok. <BR>Could anyone tell me what's the difference for alsasink to process
audio packets and files? How to fix the above issue? thank you very
much!<BR> <BR>Best Regards!<BR>Shenhong WANG<BR><BR>
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