<div></div>
<div>
Hi Jan,</div><div><br></div><div>Thank you, that is definitely what I need. I have gotten that pipeline working successfully, but now when I play it (via ffplay), it sounds comically slow/distorted.</div><div><br></div><div>I am now using the pipeline:</div><div><br></div><div>gst-launch-1.0 -m filesrc location=myfile.wav ! wavparse ! audioconvert ! rtpL16pay ! udpsink host=127.0.0.1 port=12008</div><div><br></div><div>I used the -v option to produce what I thought was a correct SDP file:</div><div><br></div><div><div>v=0</div><div>o=root IN IP4 127.0.0.1</div><div>c=IN IP4 127.0.0.1</div><div>s=My Name</div><div>m=audio 12008 RTP/AVP 96</div><div>a=rtpmap:96 L16/44100</div><div>a=fmtp:96 media=audio; clock-rate=44100; encoding-name=L16; channels=2;</div><div><br></div><div>I am playing the sound with:</div><div><br></div><div>ffplay -i stream.sdp -protocol_whitelist file,udp,rtp<br></div><div><br></div><div>Ffplay does open, and the sound resembles the original song, but it is very slowed down/distorted. </div><div><br></div><div>Ffplay sees:</div><div><br></div><div><div>bitrate: 705 kb/s</div><div> Stream #0:0: Audio: pcm_s16be, 44100 Hz, 1 channels, s16, 705 kb/s</div></div><div><br></div><div>(not sure if that will help)</div><div><br></div><div>I have been trying to research what is wrong here but I am not sure what part of this I’ve messed up. Any help would be much appreciated. Thank you!</div><div><br></div><div>Josh</div><br>
</div>
<div id="psignature">
<div><br></div>
<div style="font-size:10px; color: #7E8F9F;"></div>
</div>
<div class="gmail_extra"><br><div class="gmail_quote"><div dir="ltr">On Fri, May 12, 2017 at 01:39 Jan Schmidt <thaytan@noraisin.net> <<a href="mailto:Jan Schmidt <thaytan@noraisin.net>">Jan Schmidt <thaytan@noraisin.net></a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><!DOCTYPE html><html><head><meta name="viewport" content="width=device-width"><style>a, pre, code, a:link, body { word-wrap: break-word !important; }</style>
</head><body>
<p>Hi,<br>
</p>
<br>
<div class="moz-cite-prefix"></div>
</body></html>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>
</div>
<div id="psignature">
<div><br>
</div>
</div>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
gstreamer-devel mailing list
<a class="moz-txt-link-abbreviated" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>
<a class="moz-txt-link-freetext" href="https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a>
</pre>
</blockquote>
<br>
</body>
</html>
</div></blockquote></div><br></div>