Why your source element has a break? do you use live source?
I suggest you pause the pipleline at the moment.
Brad
________________________________
From: Shenhong Wang [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 18, 2008 4:39 PM
To: Zhao Bin-E6223C; gstreamer-embedded@lists.sourceforge.net
Subject: RE: [gst-embedded] Question on gst_plugin alsasink
Brad,
thanks!
At the moment I didn't put any timestamp on the frames(buffer) yet.
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!
Best Regards!
Shenhong WANG
________________________________
Subject: RE: [gst-embedded] Question on gst_plugin alsasink
Date: Wed, 18 Jun 2008 15:01:06 +0800
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED];
gstreamer-embedded@lists.sourceforge.net
Hi, your issue seems that audio frame is delayed when arrivering
alsasink.
basesink will drop the delayed buffer and you could'nt hear any
sound.
Please check your timestamp of buffer.
Brad
________________________________
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Shenhong Wang
Sent: Wednesday, June 18, 2008 2:21 PM
To: gstreamer-embedded@lists.sourceforge.net
Subject: [gst-embedded] Question on gst_plugin alsasink
Dear all,
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.
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.
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!
Best Regards!
Shenhong WANG
________________________________
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