<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"> <HTML><HEAD> <META http-equiv=Content-Type content="text/html; charset=gb2312"> <META content="MSHTML 6.00.2900.3157" name=GENERATOR> <STYLE>@font-face { font-family: 宋体; } @font-face { font-family: Verdana; } @font-face { font-family: @宋体; } @page Section1 {size: 595.3pt 841.9pt; margin: 72.0pt 90.0pt 72.0pt 90.0pt; layout-grid: 15.6pt; } P.MsoNormal { TEXT-JUSTIFY: inter-ideograph; FONT-SIZE: 10.5pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"; TEXT-ALIGN: justify } LI.MsoNormal { TEXT-JUSTIFY: inter-ideograph; FONT-SIZE: 10.5pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"; TEXT-ALIGN: justify } DIV.MsoNormal { TEXT-JUSTIFY: inter-ideograph; FONT-SIZE: 10.5pt; MARGIN: 0cm 0cm 0pt; FONT-FAMILY: "Times New Roman"; TEXT-ALIGN: justify } A:link { COLOR: blue; TEXT-DECORATION: underline } SPAN.MsoHyperlink { COLOR: blue; TEXT-DECORATION: underline } A:visited { COLOR: purple; TEXT-DECORATION: underline } SPAN.MsoHyperlinkFollowed { COLOR: purple; TEXT-DECORATION: underline } SPAN.EmailStyle17 { FONT-WEIGHT: normal; COLOR: windowtext; FONT-STYLE: normal; FONT-FAMILY: Verdana; TEXT-DECORATION: none; mso-style-type: personal-compose } DIV.Section1 { page: Section1 } UNKNOWN { FONT-SIZE: 10pt } BLOCKQUOTE { MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px; MARGIN-LEFT: 2em } OL { MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px } UL { MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px } </STYLE> </HEAD> <BODY style="FONT-SIZE: 10pt; FONT-FAMILY: verdana"> <DIV><FONT face=Verdana color=#000080 size=2> <DIV><FONT face=Verdana color=#000080 size=2><FONT color=#000000>Hi Joyious,</FONT></FONT></DIV> <DIV> </DIV> <DIV>I think the problem lies in that audioresample fails to convert from 44100 hz to 48000 hz. Your oss driver does not support 44100, but the rate of 1.mp3 is 44100. So the audioresample should resample.</DIV> <DIV>But according to your output, rate of audioresample's source pad is different with the one of osssink's sink pad. The former is 0, while the latter is 48000. </DIV> <DIV>I test the pipleline on my PC using osssink. It can work well.</DIV> <DIV> </DIV> <DIV>my command line:</DIV> <DIV>gst-launch-0.10 -v filesrc location= /home/jianjun/206851.mp3 ! mad ! audioconvert ! audioresample ! osssink</DIV> <DIV> </DIV> <DIV>And its output:</DIV> <DIV><BR>Setting pipeline to PAUSED ...<BR>Pipeline is PREROLLING ...<BR>/pipeline0/mad0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2<BR>/pipeline0/audioconvert0.src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)44100<BR>/pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)32, depth=(int)32, rate=(int)44100, channels=(int)2<BR>/pipeline0/audioresample0.src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)44100<BR>/pipeline0/audioresample0.sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)44100<BR>/pipeline0/osssink0.sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)44100<BR>Pipeline is PREROLLED ...<BR>Setting pipeline to PLAYING ...<BR>New clock: GstAudioSinkClock</DIV> <DIV> </DIV> <DIV><FONT face=Verdana color=#000080 size=2></FONT> </DIV> <DIV><FONT color=#000080>Regards,</FONT></DIV> <DIV><FONT face=Verdana color=#000080 size=2>Jianjun</FONT></DIV></FONT></DIV> <DIV><FONT face=Verdana color=#000080 size=2></FONT> </DIV> <DIV><FONT face=Verdana color=#000080 size=2></FONT> </DIV> <DIV><FONT face=Verdana color=#c0c0c0 size=2>2007-09-19 </FONT></DIV><FONT face=Verdana color=#000080 size=2> <HR style="WIDTH: 122px; HEIGHT: 2px" align=left SIZE=2> </FONT> <DIV><FONT face=Verdana color=#c0c0c0 size=2><SPAN>jianjun.yang.cn</SPAN> </FONT></DIV><FONT face=Verdana color=#000080 size=2> <HR> </FONT> <DIV><FONT face=Verdana size=2><STRONG>发件人：</STRONG> 何慧敏 </FONT></DIV> <DIV><FONT face=Verdana size=2><STRONG>发送时间：</STRONG> 2007-09-19  15:52:35 </FONT></DIV> <DIV><FONT face=Verdana size=2><STRONG>收件人：</STRONG> gstreamer-devel </FONT></DIV> <DIV><FONT face=Verdana size=2><STRONG>抄送：</STRONG> </FONT></DIV> <DIV><FONT face=Verdana size=2><STRONG>主题：</STRONG> [gst-devel] Problems when using osssink in ARM </FONT></DIV> <DIV><FONT face=Verdana size=2></FONT> </DIV> <DIV><FONT face=Verdana size=2> <DIV>Hi all,<BR><BR>Now I have ported the gstreamer to ARM 11, and there is a OSS driver for this ARM board. So I am trying to make this sound workI have installed the plugin for ossaudio.But when i trying to play some media file, no sound came out, and the screen just show the PLAYING message ..<BR><BR>some addtional messages:<BR>the osssink does play the audiotestsrc,and sounds a single tone;<BR>the audioconvert works fine when "gst-launch -v filesrc location="/usr/local/bin/1.mp3" ! flump3dec ! audioconvert ! wavenc ! filesink location="/usr/local/bin/1.wav" ",I can hear the wav file on my PC,so the decodec is fine;<BR>even no sound comes out when playing, but there are some noises when pipeline is just prerolling just before PLAYING.<BR><BR><BR>Here are the command and output:<BR>***********************************************************************************************************<BR>mx31# gst-launch -v filesrc location="/usr/local/bin/1.mp3" ! flump3dec ! audioconvert ! audioresample ! osssink<BR>Setting pipeline to PAUSED ...<BR><BR>MXC Enable Codec(write)<BR>Feb 6 12:04:07 freescale user.warn kernel: <BR>Feb 6 12:04:07 freescale user.warn kernel: MXC Enable Codec(write)<BR>Pipeline is PREROLLING ...<BR>/pipeline0/flump3dec0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2<BR>/pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(i2<BR>/pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(2<BR>/pipeline0/audioresample0.src: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)0<BR>/pipeline0/audioresample0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=2<BR>/pipeline0/osssink0.sink: caps = audio/x-raw-int, width=(int)16, depth=(int)16, signed=(boolean)true, endianness=(int)1234, channels=(int)2, rate=(int)48000<BR>Pipeline is PREROLLED ...<BR>Setting pipeline to PLAYING ...<BR>New clock: GstAudioSinkClock<BR>mxc_audio_output_block: count = 1024 <BR>Feb 6 12:04:37 freescale user.warn kernel: mxc_audio_output_block: count = 1024 <BR><BR>*************************************************************************************************************<BR><BR>So I'm studying the code of osssink although I am a totally newcomer to this gstreamer and Linux things.<BR><BR>Many Thanks,<BR><BR>Joyious<BR></DIV> <DIV> </DIV> <DIV> </DIV> <DIV> </DIV><BR><BR> <HR> <FONT style="FONT-SIZE: 12px; LINE-HEIGHT: 15px">杀70万种木马病毒，</FONT><A style="FONT-SIZE: 12px; COLOR: blue; LINE-HEIGHT: 15px; TEXT-DECORATION: underline" href="http://pro.163.com/event.ng/Type=click&FlightID=96648&AdID=98138&TargetID=635&Values=31,43,51,60,72,82,91,100,110,312,330,332,499,587,702,733,734&Redirect=http://adcn.doubleclick.net/clk;134682177;20226578;k%3Fhttp://www.rising.com.cn/2008/trial/index.htm">瑞星2008版免费</A> </FONT></DIV></BODY></HTML>