Could not capture AAC audio in mp4mux properly, help !!!
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Hi,
I used the following pipeline: gst-launch alsasrc ! faac, bitrate=48000, profile=2 ! mp4mux ! filesink location=test.m4a The file was generated, but it could not be played back by gstreamer using the following pipeline: gst-launch filesrc location=test.m4a ! decodebin ! autoaudiosink Following messages are generated. It seems that the file was not generated properly. Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0: This file is invalid and cannot be played. Additional debug info: qtdemux.c(1447): gst_qtdemux_loop_state_header (): /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0: Header atom 'mdat' has empty length ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... I checked the binary, and the length field of mdat is indeed 0, and I could not find any atoms, like moov, mvhd, etc., that describe the media. Is this the right pipeline to capture audio in MP4 file? Thanks!! Yiliang |
Re: Could not capture AAC audio in mp4mux properly, help !!!
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Set num-buffers property in alsasrc . On Wed, Oct 7, 2009 at 12:37 PM, Yiliang Bao <[hidden email]> wrote:
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Re: Could not capture AAC audio in mp4mux properly, help !!!
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On Wed, Oct 7, 2009 at 5:05 AM, sudarshan bisht <[hidden email]> wrote:
Or use "gst-launch -e". The problem here is that the pipeline is not properly finished (receives an EOS event), using -e will force an EOS when you press ctrl+c. Using num-buffers will make the alsasrc push an EOS when the number of buffers is generated.
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In reply to this post by yiliang
Hi all, is tehre an example on java based rtsp video player? I want to integrate it into my applet phone. Basically I want to contact a media server to get the stream.
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In reply to this post by Thiago Sousa Santos-2
Thanks, Thiago and Sudarshan. I tested both approaches. "-e" works
perfectly, but setting num-buffers does not work though. On Wed, Oct 7, 2009 at 3:51 AM, thiagoss <[hidden email]> wrote: > > > On Wed, Oct 7, 2009 at 5:05 AM, sudarshan bisht <[hidden email]> > wrote: >> >> Set num-buffers property in alsasrc . > > Or use "gst-launch -e". The problem here is that the pipeline is not > properly finished (receives an EOS event), using -e will force an EOS when > you press ctrl+c. Using num-buffers will make the alsasrc push an EOS when > the number of buffers is generated. > >> >> >> >> On Wed, Oct 7, 2009 at 12:37 PM, Yiliang Bao <[hidden email]> wrote: >>> >>> Hi, >>> >>> I used the following pipeline: >>> >>> gst-launch alsasrc ! faac, bitrate=48000, profile=2 ! mp4mux ! >>> filesink >>> location=test.m4a >>> >>> The file was generated, but it could not be played back by gstreamer >>> using >>> the following pipeline: >>> >>> gst-launch filesrc location=test.m4a ! decodebin ! autoaudiosink >>> >>> Following messages are generated. It seems that the file was not >>> generated >>> properly. >>> >>> Setting pipeline to PAUSED ... >>> Pipeline is PREROLLING ... >>> ERROR: from element >>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0: This >>> file is invalid and cannot be played. >>> Additional debug info: >>> qtdemux.c(1447): gst_qtdemux_loop_state_header (): >>> /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstQTDemux:qtdemux0: >>> Header atom 'mdat' has empty length >>> ERROR: pipeline doesn't want to preroll. >>> Setting pipeline to NULL ... >>> Freeing pipeline ... >>> >>> I checked the binary, and the length field of mdat is indeed 0, and I >>> could >>> not find any atoms, like moov, mvhd, etc., that describe the media. >>> >>> Is this the right pipeline to capture audio in MP4 file? Thanks!! >>> >>> Yiliang >>> >>> -- >>> View this message in context: >>> http://www.nabble.com/Could-not-capture-AAC-audio-in-mp4mux-properly%2C-help-%21%21%21-tp25781565p25781565.html >>> Sent from the GStreamer-devel mailing list archive at Nabble.com. >>> >>> >>> >>> ------------------------------------------------------------------------------ >>> Come build with us! The BlackBerry(R) Developer Conference in SF, CA >>> is the only developer event you need to attend this year. Jumpstart your >>> developing skills, take BlackBerry mobile applications to market and stay >>> ahead of the curve. Join us from November 9 - 12, 2009. Register now! >>> http://p.sf.net/sfu/devconference >>> _______________________________________________ >>> gstreamer-devel mailing list >>> [hidden email] >>> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> >> >> >> -- >> Regards, >> >> Sudarshan Bisht >> >> >> ------------------------------------------------------------------------------ >> Come build with us! The BlackBerry(R) Developer Conference in SF, CA >> is the only developer event you need to attend this year. Jumpstart your >> developing skills, take BlackBerry mobile applications to market and stay >> ahead of the curve. Join us from November 9 - 12, 2009. Register now! >> http://p.sf.net/sfu/devconference >> _______________________________________________ >> gstreamer-devel mailing list >> [hidden email] >> https://lists.sourceforge.net/lists/listinfo/gstreamer-devel >> > > > > -- > Thiago Sousa Santos > > ------------------------------------------------------------------------------ > Come build with us! The BlackBerry(R) Developer Conference in SF, CA > is the only developer event you need to attend this year. Jumpstart your > developing skills, take BlackBerry mobile applications to market and stay > ahead of the curve. Join us from November 9 - 12, 2009. Register now! > http://p.sf.net/sfu/devconference > _______________________________________________ > gstreamer-devel mailing list > [hidden email] > https://lists.sourceforge.net/lists/listinfo/gstreamer-devel > > ------------------------------------------------------------------------------ Come build with us! The BlackBerry(R) Developer Conference in SF, CA is the only developer event you need to attend this year. Jumpstart your developing skills, take BlackBerry mobile applications to market and stay ahead of the curve. Join us from November 9 - 12, 2009. Register now! http://p.sf.net/sfu/devconference _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.sourceforge.net/lists/listinfo/gstreamer-devel |
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In reply to this post by Thiago Sousa Santos-2
Thanks for your simple guilding.
I use the option (-e) And Confirm the normal end of recording and completing atoms (mdat, moov).... If User use num-buffers, User Application has a restriction of recording time... So, User might use this num-buffers cap. |
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