Hi Wes,<br><br>I see from your next post you've made a few progresses with alaw which is, (maybe) unfortunately, a codec working @ quite low bitrates. If it's enough for you to work at such levels you may want to give a try to g729 as well.<br>
<br><div class="gmail_quote">On Tue, Oct 19, 2010 at 5:18 PM, Wes Miller <span dir="ltr"><<a href="mailto:wmiller@sdr.com">wmiller@sdr.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
Marco,<br>
<br>
Better, still not quite right.<br>
<br>
Removing audioconvert and audioresample on both sender and receiver seem to<br>
have little or no effect, so they are now out.<br>
<br>
Pulsesink is working on the receiver (my Linux workstation/host). I can use<br>
pulsesrc on the sender wince Ti/RidgeRun don't seem to include the pulse<br>
stuff in their ports of gst. I keep eading about alsa hardware on the<br>
Leopardboard...???<br></blockquote><div><br>I just suggest you to fix the samplerate between alsasrc and the encoder with fixed caps, something like:<br><br>gst-launch-0.10 alsasrc ! "audio/x-raw-int, rate=44100" ! pulsesink<br>
<br>at least you'll know which sample rate you're working at ;)<br> </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
I used fakesink to get the sender caps (from fakesink0:Gstpad:sink) and I<br>
notice that the ssrc, clock-base and seqnum change every time I run the<br>
pipeline.<br></blockquote><div><br>They're expected to change each time. Some references:<br><br><a href="http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README">http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README</a><br>
</div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
If the clock-base is different each time I start the sender, how can the<br>
receiver ever actually match the sender?<br>
<br></blockquote><div><br>as clock-base is the (stream) initial time, it's expected to change slightly each time (I guess mainly because of source buffering/processing jitter). <br> </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Is there a tcp-ish way to pass the caps to the receiver and insert them in<br>
the receiver pipeline? (sounds like a great, first, element writing project,<br>
doesn't it?)<br></blockquote><div><br>yes, there are plenty of standard ways. SIP is an example, but also RTSP and MMS. All of those are excellently implmemented in many gstreamer -based applications and elements.<br>
</div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
I've tried to find out what ssrc is/are and can't find a description. So<br>
what is it? Does it matter?<br></blockquote><div><br>SSRC is the stream identification used to distinguish it among different communications about two endpoints (thinki about a videocall: audio and video are separate streams). More references here:<br>
<br><a href="http://www.ietf.org/rfc/rfc3550.txt">http://www.ietf.org/rfc/rfc3550.txt</a><br><br>Regards<br> </div><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<br>
As ever, many thanks,<br>
<br>
Wes<br>
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