HI All,
We are using the gstreamer for streaming the audio and capture the audio .we are using packages mentioned below
gst-plugins-good-0.10.14, gst-plugins-base-0.10.23, gstreamer-0.10.23, gst-plugins-bad-0.10.13
Streaming pipeline:
gst-launch alsasrc ! mulawenc ! rtppcmupay ! udpsink host=192.168.1.101 port=5555
Audio capture pipeline:
gst-launch-0.10 udpsrc port=5555 caps="application/x-rtp,media=(string)audio, clock-rate=(int)8000, encoding-name=(string)PCMU, payload=(int)0" ! rtppcmudepay ! mulawdec ! alsasink sync=false
sometimes we are getting warning mentioned below
../../../../src/gst-libs/gst/audio/gstbaseaudiosrc.c(807): gst_base_audio_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 80 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.
Thanks and Regards,
Arasu
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