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<p>Hi,<br>
</p>
<br>
<div class="moz-cite-prefix">On 12/05/17 14:43, Josh Dickson wrote:<br>
</div>
<blockquote type="cite"
cite="mid:59153b1d4c8e1a0000a6e2f5@polymail.io">
<div>
Hi,</div>
<div><br>
</div>
<div>I am sorry in advance if this is not the correct place to ask
a question…</div>
<div><br>
</div>
<div>I am trying to convert a high-quality WAV file to RTP stream.
I am successfully streaming with:</div>
<div><br>
</div>
<div>gst-launch-1.0 -m filesrc location=myfile.wav ! wavparse !
audioconvert ! audioresample ! alawenc ! rtppcmapay ! udpsink
host=127.0.0.1 port=12000</div>
</blockquote>
<br>
alaw is 8-bit @ 8khz and will generally sound awful for anything
except speech. Try rtpL16pay for 16-bit CD quality audio.<br>
<br>
Cheers,<br>
Jan.<br>
<br>
<blockquote type="cite"
cite="mid:59153b1d4c8e1a0000a6e2f5@polymail.io">
<div><br>
</div>
<div>I can then check the RTP stream from ffmpeg, which shows that
is is 64 kb/s, pct_alaw, 8000 Hz, 1 channel, s16.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>I would greatly appreciate a pointer on how I might be able
to do this. Thank you!</div>
<div><br>
</div>
<div>Josh</div>
<div><br>
</div>
<div><br>
</div>
<div><br>
<br>
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</blockquote>
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