[gst-embedded] Question on gst_plugin alsasink

Wed Jun 18 22:08:19 PDT 2008

please check queue signals "underrun" "overrun" ....

Zhao Liang
________________________________

From: Shenhong Wang [mailto:qch1688 at hotmail.com] 
Sent: Thursday, June 19, 2008 9:47 AM
To: Zhao Bin-E6223C; Zhao Liang-E3423C;
gstreamer-embedded at lists.sourceforge.net
Subject: RE: [gst-embedded] Question on gst_plugin alsasink

Hi, Brad or Zhao Liang:
Is it possible for you to publish an example - how to post a message to
bus and pause/play pipeline? thanks a lot!

Best Regards!
Shenhong

________________________________

	Subject: RE: [gst-embedded] Question on gst_plugin alsasink
	Date: Wed, 18 Jun 2008 17:08:09 +0800
	From: binzhao at motorola.com
	To: qch1688 at hotmail.com; E3423C at motorola.com;
gstreamer-embedded at lists.sourceforge.net

	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 ().

	Please check code in gstbaseaudiosink.c and gstaudiosink.c

	i remember the sig_write is lower than sig_done,sink will drop
the buffer.

________________________________

	From: Shenhong Wang [mailto:qch1688 at hotmail.com] 
	Sent: Wednesday, June 18, 2008 5:05 PM
	To: Zhao Bin-E6223C; Zhao Liang-E3423C;
gstreamer-embedded at lists.sourceforge.net
	Subject: RE: [gst-embedded] Question on gst_plugin alsasink

	Thanks! Brad.
	However I use two queues for audio and video separately but one
pipeline. So it would be impossible for me to pause the pipeline?
because the application can play video very well even the audio is
blocked. 
	Why the alsasink will drop all packets(frames) after a break or
so? thanks again

	Shenhong

________________________________

		Subject: RE: [gst-embedded] Question on gst_plugin
alsasink
		Date: Wed, 18 Jun 2008 16:55:38 +0800
		From: binzhao at motorola.com
		To: E3423C at motorola.com; qch1688 at hotmail.com;
gstreamer-embedded at lists.sourceforge.net

		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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