<div> Hi, Now I'am considering use that mad plugin, but i do not think it's the problem of the decoder. As I mentioned before, I can convert mp3 files to wav files use this decoder (using the board)and can hear the converted wav in my pc(for my board does not play any media files).At the same time i can not hear the wav file in my arm board either using the wavparse decoder. Anyway, i'll try the mad later, any more suggestion ? Many thanks Joyious</div><div> </div><div ></div> 婓2007-09-20ㄛpritesh <digitaldsp@excite.com> 迡耋ㄩ <blockquote id="isReplyContent" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"><table bgcolor="white" border="0" cellpadding="3" cellspacing="0" width="100%"><tbody><tr valign="top"><td width="100%"> hi, can u try playing your mp3 file with mad plugin , i think there is some problem with flump3dec plugin on arm11 . it never worked on my arm board also mad works fine on arm11.(freescale) u can also try ffdec_mp3 plugin from gst-ffmpeg on arm11 --- On Wed 09/19, Joyious He < <a href="mailto:joyious@163.com">joyious@163.com</a> > wrote: <blockquote style="border-left: 2px solid orange; padding-left: 7px; margin-left: 7px;">From: Joyious He [mailto: <a href="mailto:joyious@163.com">joyious@163.com</a>] To: <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> Date: Thu, 20 Sep 2007 09:50:08 +0800 (CST) Subject: Re: [gst-devel] Problems when using osssink in ARM Hi Jianjun, Here is the correct output,the previous one is copied from terminal, and there is someything shows wrong: /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 /pipeline0/audioconvert0.src: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /pipeline0/audioconvert0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /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)48000 /pipeline0/audioresample0.sink: caps = audio/x-raw-int, endianness=(int)1234, signed=(boolean)true, width=(int)16, depth=(int)16, rate=(int)44100, channels=(int)2 /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 New clock: GstAudioSinkClock so here the audio resample src seems to be accord to my osssink, but still the same, no sound came out.what should i do next ? ??2007-09-19??"jianjun.yang.cn" <jianjun.yang.cn ?="" br=""> </jianjun.yang.cn><blockquote id="isReplyContent" style="border-left: 1px solid rgb(204, 204, 204); margin: 0px 0px 0px 0.8ex; padding-left: 1ex;"> Hi Joyious, 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. 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. I test the pipleline on my PC using osssink. It can work well. my command line: gst-launch-0.10 -v filesrc location= /home/jianjun/206851.mp3 ! mad ! audioconvert ! audioresample ! osssink And its output: Setting pipeline to PAUSED ... Pipeline is PREROLLING ... /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 /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 /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 /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 /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 /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 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock Regards, Jianjun 2007-09-19 <hr style="width: 122px; height: 2px;" align="left" size="2"> jianjun.yang.cn <hr> ﹞??????? ?????? ﹞????㊣???? 2007-09-19 15:52:35 ???????? gstreamer-devel ?????? ?¯???? [gst-devel] Problems when using osssink in ARM Hi all, 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 .. some addtional messages: the osssink does play the audiotestsrc,and sounds a single tone; 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; even no sound comes out when playing, but there are some noises when pipeline is just prerolling just before PLAYING. Here are the command and output: *********************************************************************************************************** mx31# gst-launch -v filesrc location="/usr/local/bin/1.mp3" ! flump3dec ! audioconvert ! audioresample ! osssink Setting pipeline to PAUSED ... MXC Enable Codec(write) Feb 6 12:04:07 freescale user.warn kernel: Feb 6 12:04:07 freescale user.warn kernel: MXC Enable Codec(write) Pipeline is PREROLLING ... /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 /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 /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 /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 /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 /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 Pipeline is PREROLLED ... Setting pipeline to PLAYING ... New clock: GstAudioSinkClock mxc_audio_output_block: count = 1024 Feb 6 12:04:37 freescale user.warn kernel: mxc_audio_output_block: count = 1024 ************************************************************************************************************* So I'm studying the code of osssink although I am a totally newcomer to this gstreamer and Linux things. Many Thanks, Joyious  <hr> ?㊣70?辰?????赤??????<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">????2008～???﹞?</a> </blockquote>  <hr> ?㊣70?辰?????赤??????<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://ad.cn.doubleclick.net/clk;134682177;20226578;k%3Fhttp://www.rising.com.cn/2008/trial/index.htm">????2008～???﹞?</a> ------------------------------------------------------------------------- This SF.net email is sponsored by: Microsoft Defy all challenges. Microsoft(R) Visual Studio 2005. http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/_______________________________________________ gstreamer-devel mailing list <a href="mailto:gstreamer-devel@lists.sourceforge.net">gstreamer-devel@lists.sourceforge.net</a> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel </blockquote></td></tr></tbody></table><hr>Join Excite! - <a href="http://www.excite.com" target="_blank">http://www.excite.com</a> The most personalized portal on the Web! </blockquote>  <hr> 伀70勀笱躂鎮瓷馮ㄛ<a style="font-size:12px;line-height:15px; color:blue; 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://ad.cn.doubleclick.net/clk;134682177;20226578;k%3Fhttp://www.rising.com.cn/2008/trial/index.htm">⻞陎2008唳轎煤</a>