[gst-devel] Audio streaming using gstreamer issue

[tamil arasu]

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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