I want to play a wave file as following
gst-launch-1.0 -v filesrc location=./mysong.wav ! wavparse ! audioconvert ! rtpL16pay ! application/x-rtp, payload=11, encoding-name=L16, clock-rate=16000, channels=1 ! udpsink host=localhost port=6000 I am getting a warning : WARNING: erroneous pipeline: could not link rtpl16pay0 to udpsink0 The warning is due to the payload. If I am removing the payload the vlc is showing error : A description in SDP format is required to receive the RTP stream. Note that rtp:// URIs cannot work with dynamic RTP payload format (96) My file information : Input File : 'mysong.wav' Channels : 2 Sample Rate : 16000 Precision : 16-bit Duration : 00:00:54.31 = 868984 samples ~ 4073.36 CDDA sectors File Size : 3.48M Bit Rate : 512k Sample Encoding: 16-bit Signed Integer PCM How to solve this problem. Thank You -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Tue, 2017-09-19 at 23:03 -0700, amsts wrote:
Hi, > I want to play a wave file as following > > gst-launch-1.0 -v filesrc location=./mysong.wav ! wavparse ! > audioconvert ! > rtpL16pay ! application/x-rtp, payload=11, encoding-name=L16, > clock-rate=16000, channels=1 ! udpsink host=localhost port=6000 > > I am getting a warning : > > WARNING: erroneous pipeline: could not link rtpl16pay0 to udpsink0 Look at the source pad template caps in gst-inspect-1.0 rtpL16pay. If you use payload type 10 or 11 that means you must use a sample rate of 44100, which will result in clock-rate=44100 on the rtp output caps. Your problem is that you specify both payload=11 and a non-44100 clock- rate. https://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rt p-parameters-1 One solution is to use an audioresample before rtpL16pay as well to convert the input sample rate to 44100. (It seems the only solution unless you wanat to use/re-encode to a different format, since the restriction/requirement for non-dynamic pt is on the VLC side according to the error message you post, if I understood it correctly). Cheers -Tim -- Tim Müller, Centricular Ltd - http://www.centricular.com Join us at the GStreamer Conference! 21-22 October 2017 in Prague, Czech Republic http://gstreamer.freedesktop.org/conference/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Hi Tim,
Thank You for the reply. I removed payload as you said but now I am getting another error and I am not able to find what it is. Can you help me out. gst-launch-1.0 filesrc location=./newsong.wav ! wavparse ! audioconvert ! rtpL16pay ! application/x-rtp, encoding-name=L16, payload=11, channels=1,layout=interleaved ! udpsink host=localhost port=6000 Setting pipeline to PAUSED ... Pipeline is PREROLLING ... ERROR: from element /GstPipeline:pipeline0/GstWavParse:wavparse0: Internal data flow error. Additional debug info: gstwavparse.c(2261): gst_wavparse_loop (): /GstPipeline:pipeline0/GstWavParse:wavparse0: streaming task paused, reason not-negotiated (-4) ERROR: pipeline doesn't want to preroll. Setting pipeline to NULL ... Freeing pipeline ... Thank You. -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
On Wed, 2017-09-20 at 07:51 -0700, amsts wrote:
Hi, > I removed payload as you said but now I am getting another error and > I am not able to find what it is. > > gst-launch-1.0 filesrc location=./newsong.wav ! wavparse ! > audioconvert ! rtpL16pay ! application/x-rtp, encoding-name=L16, > payload=11, channels=1,layout=interleaved ! udpsink host=localhost > port=6000 > > streaming task paused, reason not-negotiated (-4) This means there is a problem with the caps/formats, something is not compatible. I'm guessing the output of wavparse is not compatible in some way (format, sample rate, number of channels) with what rtpL16pay requires when forced to those output caps. Add ... ! wavparse ! audioresample ! audioconvert ! rtpL16pay ! .. Cheers -Tim -- Tim Müller, Centricular Ltd - http://www.centricular.com Join us at the GStreamer Conference! 21-22 October 2017 in Prague, Czech Republic http://gstreamer.freedesktop.org/conference/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Thanks a lot, Tim.
It helped me a lot. But, when I see the debug data. I can see that the clock rate is 44100. But in my requirement, the receiver should receive 16000hz frequency, linear pcm 16 bit and payload should be 11. With above change, I am completing only two requirements I can not change the receiver settings. Is it possible to satisfy all requirements? Anyway thanks once again. -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
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