Using GStreamer to Convert WAV -> RTP Results in Low-Quality RTP Stream

[Josh Dickson]

Hi,

I am sorry in advance if this is not the correct place to ask a question…

I am trying to convert a high-quality WAV file to RTP stream. I am successfully streaming with:

gst-launch-1.0 -m filesrc location=myfile.wav ! wavparse ! audioconvert ! audioresample ! alawenc ! rtppcmapay ! udpsink host=127.0.0.1 port=12000

I can then check the RTP stream from ffmpeg, which shows that is is 64 kb/s, pct_alaw, 8000 Hz, 1 channel, s16.

My WAV file is much higher quality than this (it is a sample of music at CD quality). I thought that the problem was with audioresample, but I have tried a number of changes and I cannot get any of them to stream correctly. Ideally the stream should be as high-quality as the WAV it’s generated from.

I would greatly appreciate a pointer on how I might be able to do this. Thank you!

Josh
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