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
She
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 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.n
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 somet
Brad,
Yes, now I am using a live source via wireless signal.
Questions:
a) I don't know when the break will happen
b) I don't know how to pause the pipeline
How to move it forward? thanks!
Best Regards!
Shenhong WANG
Subject: RE: [gst-embedded] Question on gst_plugin alsasinkDate: Wed, 18 Jun
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:[EMAIL PROTECTED]
Sent: Wednesday, Jun
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: [EMAIL PROTECTED]: [E
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.
Fr
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 a
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
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 alsasinkDate: Wed, 18 Jun
2008 17:08:09 +0800From: [EMAIL PROTECTED]: [EMAIL PRO
please check queue signals "underrun" "overrun"
Zhao Liang
From: Shenhong Wang [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 19, 2008 9:47 AM
To: Zhao Bin-E6223C; Zhao Liang-E3423C;
gstreamer-embedded@lists.sourceforge.net
Subject: RE: [gst-embedded] Questi
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