<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HTML><HEAD> <META http-equiv=Content-Type content="text/html; charset=us-ascii"> <STYLE>.hmmessage P { PADDING-RIGHT: 0px; PADDING-LEFT: 0px; PADDING-BOTTOM: 0px; MARGIN: 0px; PADDING-TOP: 0px } BODY.hmmessage { FONT-SIZE: 10pt; FONT-FAMILY: Tahoma } </STYLE> <META content="MSHTML 6.00.2900.3314" name=GENERATOR></HEAD> <BODY class=hmmessage> <DIV dir=ltr align=left><FONT face=Arial color=#0000ff></FONT> </DIV> <DIV><FONT face="Courier New" color=#0000ff>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 ().</FONT></DIV> <DIV><FONT face="Courier New" color=#0000ff></FONT> </DIV> <DIV><SPAN class=652390609-18062008><FONT face="Courier New" color=#0000ff>Please check code in gstbaseaudiosink.c and gstaudiosink.c</FONT></SPAN></DIV> <DIV><SPAN class=652390609-18062008><FONT face="Courier New" color=#0000ff></FONT></SPAN> </DIV> <DIV><SPAN class=652390609-18062008><FONT face="Courier New" color=#0000ff>i remember the sig_write is lower than sig_done,sink will drop the buffer.</FONT></SPAN></DIV><BR> <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left> <HR tabIndex=-1> <FONT face=Tahoma><B>From:</B> Shenhong Wang [mailto:qch1688@hotmail.com] <BR><B>Sent:</B> Wednesday, June 18, 2008 5:05 PM<BR><B>To:</B> Zhao Bin-E6223C; Zhao Liang-E3423C; gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> RE: [gst-embedded] Question on gst_plugin alsasink<BR></FONT><BR></DIV> <DIV></DIV>Thanks! Brad.<BR>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. <BR>Why the alsasink will drop all packets(frames) after a break or so? thanks again<BR> <BR>Shenhong<BR><BR><BR><BR><BR> <BLOCKQUOTE> <HR id=EC_stopSpelling> Subject: RE: [gst-embedded] Question on gst_plugin alsasink<BR>Date: Wed, 18 Jun 2008 16:55:38 +0800<BR>From: binzhao@motorola.com<BR>To: E3423C@motorola.com; qch1688@hotmail.com; gstreamer-embedded@lists.sourceforge.net<BR><BR> <META content="Microsoft SafeHTML" name=Generator> <STYLE>.ExternalClass .EC_hmmessage P { PADDING-RIGHT: 0px; PADDING-LEFT: 0px; PADDING-BOTTOM: 0px; PADDING-TOP: 0px } .ExternalClass BODY.EC_hmmessage { FONT-SIZE: 10pt; FONT-FAMILY: Tahoma } </STYLE> <DIV dir=ltr align=left><FONT face=Arial color=#0000ff></FONT> </DIV> <DIV> </DIV> <DIV align=left><SPAN lang=EN-US><FONT face=Arial color=#0000ff><SPAN class=EC_592535008-18062008>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 </SPAN></FONT></SPAN><SPAN lang=EN-US><FONT face=Arial color=#0000ff><SPAN class=EC_592535008-18062008>the message to pause the pipeline. </SPAN></FONT></SPAN></DIV> <DIV align=left><SPAN lang=EN-US><FONT face=Arial color=#0000ff><SPAN class=EC_592535008-18062008></SPAN></FONT></SPAN> </DIV> <DIV align=left><SPAN lang=EN-US><FONT face=Arial color=#0000ff><SPAN class=EC_592535008-18062008>if higher water mark, you can use the same mechanism.</DIV> <DIV align=left><SPAN class=EC_169584808-18062008><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV align=left><SPAN class=EC_169584808-18062008><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV> <DIV align=left><SPAN class=EC_169584808-18062008><FONT face=Arial color=#0000ff></FONT></SPAN> </DIV></SPAN></FONT></SPAN><BR> <DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left> <HR> <FONT face=Tahoma><B>From:</B> gstreamer-embedded-bounces@lists.sourceforge.net [mailto:gstreamer-embedded-bounces@lists.sourceforge.net] <B>On Behalf Of </B>Zhao Liang-E3423C<BR><B>Sent:</B> Wednesday, June 18, 2008 4:49 PM<BR><B>To:</B> Shenhong Wang; gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> Re: [gst-embedded] Question on gst_plugin alsasink<BR></FONT><BR></DIV> <DIV></DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3>Hi <SPAN class=EC_999344508-18062008>she</SPAN>nhong<SPAN class=EC_999344508-18062008>,</SPAN></FONT></FONT></FONT></DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3><SPAN class=EC_999344508-18062008></SPAN></FONT></FONT></FONT> </DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3><SPAN class=EC_999344508-18062008>A simply solution you can try.</SPAN></FONT></FONT></FONT></DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3><SPAN class=EC_999344508-18062008></SPAN></FONT></FONT></FONT> </DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3><SPAN class=EC_999344508-18062008>Put a queue before alsasink, when queue is dry, pause pipeline, and restart pipeline when queue bufferred enough data.</SPAN></FONT></FONT></FONT></DIV> <DIV dir=ltr align=left><FONT face="Courier New"><FONT color=#0000ff><FONT size=3><SPAN class=EC_999344508-18062008></SPAN></FONT></FONT></FONT> </DIV> <DIV><FONT face="Courier New" color=#0000ff size=3></FONT> </DIV> <DIV class=EC_Section1> <P class=EC_MsoNormal align=left><B><SPAN lang=EN-US style="FONT-SIZE: 10pt; COLOR: red; FONT-FAMILY: 'Courier New'">Best Regards<BR>Zhao <SPAN class=EC_SpellE>Liang</SPAN> </SPAN></B></P></DIV> <DIV class=EC_OutlookMessageHeader lang=en-us dir=ltr align=left> <HR> <FONT face=Tahoma size=2><B>From:</B> Shenhong Wang [mailto:qch1688@hotmail.com] <BR><B>Sent:</B> Wednesday, June 18, 2008 4:44 PM<BR><B>To:</B> Zhao Liang-E3423C; gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> RE: [gst-embedded] Question on gst_plugin alsasink<BR></FONT><BR></DIV> <DIV></DIV>Hi, Zhao Liang:<BR>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. <BR>It looks the aacdec is slower than the expectation from alsasink.If so, how to fix the issue? thanks!<BR> <BR>best Regards!<BR>Shenhong<BR> <BR> <BR><BR><BR><BR><BR> <BR> <BLOCKQUOTE> <HR id=EC_EC_stopSpelling> Subject: RE: [gst-embedded] Question on gst_plugin alsasink<BR>Date: Wed, 18 Jun 2008 14:29:27 +0800<BR>From: E3423C@motorola.com<BR>To: qch1688@hotmail.com; gstreamer-embedded@lists.sourceforge.net<BR><BR> <STYLE>.ExternalClass .EC_hmmessage P { PADDING-RIGHT: 0px; PADDING-LEFT: 0px; PADDING-BOTTOM: 0px; PADDING-TOP: 0px } .ExternalClass BODY.EC_hmmessage { FONT-SIZE: 10pt; FONT-FAMILY: Tahoma } </STYLE> <DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff size=3><SPAN class=EC_EC_329532206-18062008>Hi Shenhong,</SPAN></FONT></DIV> <DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff size=3><SPAN class=EC_EC_329532206-18062008></SPAN></FONT> </DIV> <DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff size=3><SPAN class=EC_EC_329532206-18062008>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 ().</SPAN></FONT></DIV> <DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff size=3><SPAN class=EC_EC_329532206-18062008></SPAN></FONT> </DIV> <DIV dir=ltr align=left><FONT face="Courier New" color=#0000ff size=3><SPAN class=EC_EC_329532206-18062008>For the rootcause, I think maybe the alsasink audiodevice buffer is too big or your aac decoder is too slow.</SPAN></FONT></DIV> <DIV><FONT face="Courier New" color=#0000ff size=3></FONT> </DIV> <DIV class=EC_EC_Section1> <P class=EC_EC_MsoNormal align=left><B><SPAN lang=EN-US style="FONT-SIZE: 10pt; COLOR: red; FONT-FAMILY: 'Courier New'">Best Regards<BR>Zhao <SPAN class=EC_EC_SpellE>Liang</SPAN></SPAN></B><BR></P></DIV> <DIV class=EC_EC_OutlookMessageHeader lang=en-us dir=ltr align=left> <HR> <FONT face=Tahoma size=2><B>From:</B> gstreamer-embedded-bounces@lists.sourceforge.net [mailto:gstreamer-embedded-bounces@lists.sourceforge.net] <B>On Behalf Of </B>Shenhong Wang<BR><B>Sent:</B> Wednesday, June 18, 2008 2:21 PM<BR><B>To:</B> gstreamer-embedded@lists.sourceforge.net<BR><B>Subject:</B> [gst-embedded] Question on gst_plugin alsasink<BR></FONT><BR></DIV> <DIV></DIV><BR>Dear all,<BR>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. <BR>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. <BR>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!<BR> <BR>Best Regards!<BR>Shenhong WANG<BR><BR> <HR> Connect to the next generation of MSN Messenger  <A href="http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline" target=_blank>Get it now! </A></BLOCKQUOTE><BR> <HR> Connect to the next generation of MSN Messenger  <A href="http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline" target=_blank>Get it now! </A></BLOCKQUOTE><BR> <HR> Connect to the next generation of MSN Messenger  <A href="http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline" target=_new>Get it now! </A></BODY></HTML>