[gst-devel] Choppy Audio over UDP

[Marco Ballesio]

Hi Wes,

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.

On Tue, Oct 19, 2010 at 5:18 PM, Wes Miller <wmiller at sdr.com> wrote:

>
> Marco,
>
> Better, still not quite right.
>
> Removing audioconvert and audioresample on both sender and receiver seem to
> have little or no effect, so they are now out.
>
> Pulsesink is working on the receiver (my Linux workstation/host).  I can
> use
> pulsesrc on the sender wince Ti/RidgeRun don't seem to include the pulse
> stuff in their ports of gst.  I keep eading about alsa hardware on the
> Leopardboard...???
>

I just suggest you to fix the samplerate between alsasrc and the encoder
with fixed caps, something like:

gst-launch-0.10 alsasrc ! "audio/x-raw-int, rate=44100" ! pulsesink

at least you'll know which sample rate you're working at ;)


>
> I used fakesink to get the sender caps (from fakesink0:Gstpad:sink) and I
> notice that the ssrc, clock-base and seqnum change every time I run the
> pipeline.
>

They're expected to change each time. Some references:

http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README


>
> If the clock-base is different each time I start the sender, how can the
> receiver ever actually match the sender?
>
>
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).


> Is there a tcp-ish way to pass the caps to the receiver and insert them in
> the receiver pipeline? (sounds like a great, first, element writing
> project,
> doesn't it?)
>

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.


>
> I've tried to find out what ssrc is/are and can't find a description.  So
> what is it? Does it matter?
>

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:

http://www.ietf.org/rfc/rfc3550.txt

Regards


>
> As ever, many thanks,
>
> Wes
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3002180.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
>
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