osssrc complains "Can't record audio fast enough"
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case 1: "gst-launch osssrc ! lame ! filesink location=xxx" gives "osssrc: Can't record audio fast enough".
case 2" "gst-launch osssrc ! audioconvert ! osssink sync=false" gives OK. Any way to generate mp3 files with osssrc? Thanks! ------------------------------------------------------------------------------ This SF.Net email is sponsored by the Verizon Developer Community Take advantage of Verizon's best-in-class app development support A streamlined, 14 day to market process makes app distribution fast and easy Join now and get one step closer to millions of Verizon customers http://p.sf.net/sfu/verizon-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: osssrc complains "Can't record audio fast enough"
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although I am unable to regenerate your test case, you might try
gst-launch osssrc ! lame ! filesink location=xxx sync=true On Fri, Dec 25, 2009 at 1:19 PM, gather bzbz <[hidden email]> wrote: case 1: "gst-launch osssrc ! lame ! filesink location=xxx" gives "osssrc: Can't record audio fast enough". -- http://www.linkedin.com/in/kapilagrawal ------------------------------------------------------------------------------ This SF.Net email is sponsored by the Verizon Developer Community Take advantage of Verizon's best-in-class app development support A streamlined, 14 day to market process makes app distribution fast and easy Join now and get one step closer to millions of Verizon customers http://p.sf.net/sfu/verizon-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
Re: osssrc complains "Can't record audio fast enough"
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In reply to this post by G.B
On Thu, 2009-12-24 at 23:49 -0800, gather bzbz wrote:
> case 1: "gst-launch osssrc ! lame ! filesink location=xxx" gives "osssrc: Can't record audio fast enough". Note that for any half-way recent version of gstreamer/gst-plugins-base the debug message that comes with that warning reads something like "Dropped x samples. This is most likely because downstream can't keep up and is consuming samples too slowly." Try putting a queue between osssrc and lame. If that doesn't fix it, then the encoder (lame) is most likely just not able to encode fast enough, ie. in real time on average, on that particular computer, in which case you might be able to configure the encoder to use a less computationally intensive algorithm/mode (see gst-inspect-0.10 lame; also have a look at the lamemp3enc element which has simpler options). If that still doesn't help, you could try to reduce the input sample rate (by adding a caps filter between source and the encoder, for example). Cheers -Tim ------------------------------------------------------------------------------ This SF.Net email is sponsored by the Verizon Developer Community Take advantage of Verizon's best-in-class app development support A streamlined, 14 day to market process makes app distribution fast and easy Join now and get one step closer to millions of Verizon customers http://p.sf.net/sfu/verizon-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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Hi Tim,
Could you please provide a sample of the said cap filter set between the src and lame? I have tried other ways, none worked :( Best, B.Z. --- On Fri, 12/25/09, Tim-Philipp Müller <[hidden email]> wrote: > From: Tim-Philipp Müller <[hidden email]> > Subject: Re: [gst-devel] osssrc complains "Can't record audio fast enough" > To: [hidden email] > Date: Friday, December 25, 2009, 8:19 AM > On Thu, 2009-12-24 at 23:49 -0800, > gather bzbz wrote: > > > case 1: "gst-launch osssrc ! lame ! filesink > location=xxx" gives "osssrc: Can't record audio fast > enough". > > Note that for any half-way recent version of > gstreamer/gst-plugins-base > the debug message that comes with that warning reads > something like > "Dropped x samples. This is most likely because downstream > can't keep up > and is consuming samples too slowly." > > Try putting a queue between osssrc and lame. If that > doesn't fix it, > then the encoder (lame) is most likely just not able to > encode fast > enough, ie. in real time on average, on that particular > computer, in > which case you might be able to configure the encoder to > use a less > computationally intensive algorithm/mode (see > gst-inspect-0.10 lame; > also have a look at the lamemp3enc element which has > simpler options). > > If that still doesn't help, you could try to reduce the > input sample > rate (by adding a caps filter between source and the > encoder, for > example). > > Cheers > -Tim > > > > ------------------------------------------------------------------------------ > This SF.Net email is sponsored by the Verizon Developer > Community > Take advantage of Verizon's best-in-class app development > support > A streamlined, 14 day to market process makes app > distribution fast and easy > Join now and get one step closer to millions of Verizon > customers > http://p.sf.net/sfu/verizon-dev2dev > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > ------------------------------------------------------------------------------ This SF.Net email is sponsored by the Verizon Developer Community Take advantage of Verizon's best-in-class app development support A streamlined, 14 day to market process makes app distribution fast and easy Join now and get one step closer to millions of Verizon customers http://p.sf.net/sfu/verizon-dev2dev _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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