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<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>
<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>
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