[gst-embedded] Question on gst_plugin alsasink

[Shenhong Wang]

Zhao Liang:Thanks! Now we use a queue before the aac decoder &alsasink. How to check the queue is empty and pause/restart pipeline? hehe...thanks!
 
Best Regards!
Shenhong


Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 16:49:08 +0800From: E3423C at motorola.comTo: qch1688 at hotmail.com; gstreamer-embedded at lists.sourceforge.net



Hi shenhong,
 
A simply solution you can try.
 
Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data.
 
 

Best RegardsZhao Liang 


From: Shenhong Wang [mailto:qch1688 at hotmail.com] Sent: Wednesday, June 18, 2008 4:44 PMTo: Zhao Liang-E3423C; gstreamer-embedded at lists.sourceforge.netSubject: RE: [gst-embedded] Question on gst_plugin alsasink
Hi, Zhao Liang: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 drops all frames(packets) from the break. The break is needed because we have more video frames and sometime the wireless signal is not good. It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks! best Regards!Shenhong   


Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun 2008 14:29:27 +0800From: E3423C at motorola.comTo: qch1688 at hotmail.com; gstreamer-embedded at lists.sourceforge.net


Hi Shenhong,
 
Your issue is very similar with the issue I even met. I think it is due to gstbaseaudiosink/gstaudiosink, it will drop the packets by gstringbuffer when read rate is bigger than write rate in ringbuffer, please see gstringbuffer.c gst_ring_buffer_commit_full ().
 
For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow.
 

Best RegardsZhao Liang


From: gstreamer-embedded-bounces at lists.sourceforge.net [mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On Behalf Of Shenhong WangSent: Wednesday, June 18, 2008 2:21 PMTo: gstreamer-embedded at lists.sourceforge.netSubject: [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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