Using GStreamer to Convert WAV -> RTP Results in Low-Quality RTP Stream
thaytan at noraisin.net
Fri May 12 05:39:28 UTC 2017
On 12/05/17 14:43, Josh Dickson wrote:
> 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
alaw is 8-bit @ 8khz and will generally sound awful for anything except
speech. Try rtpL16pay for 16-bit CD quality audio.
> 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!
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
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