Home Recieving audio stream with Gstreamer results in reason not negotiated error
Reply: 0

Recieving audio stream with Gstreamer results in reason not negotiated error

user5873
1#
user5873 Published in May 28, 2018, 10:02 am

I would like to stream audio data from MIC with Gstreamer. However I could not play MIC audio with rx. How can I play audio stream from MIC input?

tx: gst-launch-1.0 -v alsasrc device="hw:0" ! decodebin ! audioconvert ! rtpL16pay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004

rx: gst-launch-1.0 udpsrc multicast-group=239.0.0.1 port=5004 caps="application/x-rtp" ! rtpL16depay ! alsasink

rx result: Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstSystemClock ERROR: from element /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: Internal data flow error. Additional debug info: ../../../../gstreamer-1.8.1/libs/gst/base/gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstUDPSrc:udpsrc0: streaming task paused, reason not-negotiated (-4) Execution ended after 0:00:00.009364000 Setting pipeline to PAUSED ... Setting pipeline to READY ... Setting pipeline to NULL ... Freeing pipeline ...

tx result is as follows.

Setting pipeline to PAUSED ... Pipeline is live and does not need PREROLL ... Setting pipeline to PLAYING ... New clock: GstAudioSrcClock /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-buffer-time = 200000 /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: actual-latency-time = 10000 /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0.GstPad:src: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink.GstProxyPad:proxypad0: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:src: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" Redistribute latency... /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:src: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:src: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstQueue:queue0.GstPad:src: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstUDPSink:udpsink0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstQueue:queue0.GstPad:sink: caps = "application/x-rtp\,\ media\=(string)audio\,\ clock-rate\=(int)44100\,\ encoding-name\=(string)L16\,\ encoding-params\=(string)2\,\ channels\=(int)2\,\ payload\=(int)96\,\ ssrc\=(uint)3961155089\,\ timestamp-offset\=(uint)725507323\,\ seqnum-offset\=(uint)20783" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0.GstPad:sink: caps = "audio/x-raw\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ format\=(string)S16BE\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstAudioConvert:audioconvert0.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstDecodePad:src_0.GstProxyPad:proxypad1: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind.GstPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstDecodeBin:decodebin0.GstGhostPad:sink: caps = "audio/x-raw\,\ format\=(string)S16LE\,\ layout\=(string)interleaved\,\ rate\=(int)44100\,\ channels\=(int)2\,\ channel-mask\=(bitmask)0x0000000000000003" /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: timestamp = 725507323 /GstPipeline:pipeline0/GstRtpL16Pay:rtpl16pay0: seqnum = 20783

I think rx pipeline is wrong, but I could not find the solution. Please tell me how to make pipeline.

PS: I tried following command, and rx play the mic audio! This means the reciever device is not able to play L16 audio?

tx: gst-launch-1.0 -v alsasrc device="hw:0" ! decodebin ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! queue ! udpsink host=239.0.0.1 auto-multicast=true port=5004

rx: gst-launch-1.0 udpsrc multicast-group=239.0.0.1 port=5004 caps="application/x-rtp, media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMA, encoding-params=(string)2, channels=(int)1, payload=(int)8" ! rtppcmadepay ! alawdec ! alsasink

You need to login account before you can post.

About| Privacy statement| Terms of Service| Advertising| Contact us| Help| Sitemap|
Processed in 0.38005 second(s) , Gzip On .

© 2016 Powered by mzan.com design MATCHINFO