[gst-embedded] Question on gst_plugin alsasink

[Zhao Bin-E6223C]

yes, you can refernce how to use queue. you can set water mark in
queue.And then post message to bus if lower than mater mark. in your
main app you can recieve the message to pause the pipeline. 
 
if higher water mark, you can use the same mechanism.
 
 
 

________________________________

From: gstreamer-embedded-bounces at lists.sourceforge.net
[mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On Behalf Of
Zhao Liang-E3423C
Sent: Wednesday, June 18, 2008 4:49 PM
To: Shenhong Wang; gstreamer-embedded at lists.sourceforge.net
Subject: Re: [gst-embedded] Question on gst_plugin alsasink


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 Regards
Zhao Liang 

________________________________

From: Shenhong Wang [mailto:qch1688 at hotmail.com] 
Sent: Wednesday, June 18, 2008 4:44 PM
To: Zhao Liang-E3423C; gstreamer-embedded at lists.sourceforge.net
Subject: 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 alsasink
	Date: Wed, 18 Jun 2008 14:29:27 +0800
	From: E3423C at motorola.com
	To: 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 Regards
	Zhao Liang
	

________________________________

	From: gstreamer-embedded-bounces at lists.sourceforge.net
[mailto:gstreamer-embedded-bounces at lists.sourceforge.net] On Behalf Of
Shenhong Wang
	Sent: Wednesday, June 18, 2008 2:21 PM
	To: gstreamer-embedded at 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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