[gst-devel] gstreamer-devel Digest, Vol 53, Issue 52
Mark Beihoffer
mbeihoffer at gmail.com
Wed Oct 20 07:04:34 CEST 2010
Thants, guise! I tried that and it worked!
But the Perl bindings are still hidden in my options.
Mark Beihoffer
Dragonfly Networks
mbeihoffer at gmail.com
mark at dragonfly-networks.com
(612)508-5128
On Tue, Oct 19, 2010 at 2:51 PM, <
gstreamer-devel-request at lists.sourceforge.net> wrote:
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> Today's Topics:
>
> 1. Re: Choppy Audio over UDP (Wes Miller)
> 2. how to create a simple "on/off" element in the pipeline ?
> (Wiktor Lisowicz)
> 3. Re: how to create a simple "on/off" element in the pipeline ?
> (Tim-Philipp M?ller)
> 4. Re: long pauses when viewing RTSP stream (Gruenke, Matt)
> 5. Re: DV capture pipeline frozen (Andoni Morales)
> 6. Re: Choppy Audio over UDP (Wes Miller)
> 7. Re: Reduce latency for a MJPEG over UDP multicast pipe
> (Arnout Vandecappelle)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 19 Oct 2010 07:18:24 -0700 (PDT)
> From: Wes Miller <wmiller at sdr.com>
> Subject: Re: [gst-devel] Choppy Audio over UDP
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <1287497904585-3002180.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
>
> 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 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.
>
> If the clock-base is different each time I start the sender, how can the
> receiver ever actually match the sender?
>
> 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?)
>
> I've tried to find out what ssrc is/are and can't find a description. So
> what is it? Does it matter?
>
> 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.
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 19 Oct 2010 17:11:28 +0200
> From: Wiktor Lisowicz <greenender at gmail.com>
> Subject: [gst-devel] how to create a simple "on/off" element in the
> pipeline ?
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID:
> <AANLkTimUXDP8-6RcRSQ2Hrqr05=1ZsuZf7cmrS6joPB+ at mail.gmail.com<1ZsuZf7cmrS6joPB%2B at mail.gmail.com>
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> Following pipeline is equivalent to mine:
> gst-launch filesrc location=foobar.mp3 ! decodebin ! tee name=a a. ! queue
> !
> audioconvert ! audioresample ! autoaudiosink a. ! queue ! fakesink
>
> Short summary: I have a mp3 player, which sends the data to tee element.
> Tee
> element copies the data to two audio sinks. So instead of stereo sound, I
> have 2x stereo sound.
>
> I would like to ocasionally switch off/on data flow to one of the two
> audiosinks. This should be doable independent of the pipeline state (could
> be PAUSED, READY, PLAYING - does not matter).
>
> Which existing element could I add between <queue> and <audio sink>, to be
> able to turn on / turn off data flow to audiosink?? Is there an element
> like
> Identity (
>
> http://www.gstreamer.net/data/doc/gstreamer/0.10.3/gstreamer-plugins/html/gstreamer-plugins-identity.html
> ),
> but with additional feature of stopping the data flow?
> -------------- next part --------------
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> ------------------------------
>
> Message: 3
> Date: Tue, 19 Oct 2010 16:32:33 +0100
> From: Tim-Philipp M?ller <t.i.m at zen.co.uk>
> Subject: Re: [gst-devel] how to create a simple "on/off" element in
> the pipeline ?
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <1287502353.20478.20.camel at zingle>
> Content-Type: text/plain; charset="UTF-8"
>
> On Tue, 2010-10-19 at 17:11 +0200, Wiktor Lisowicz wrote:
>
> > I would like to ocasionally switch off/on data flow to one of the two
> > audiosinks. This should be doable independent of the pipeline state
> > (could be PAUSED, READY, PLAYING - does not matter).
> >
> > Which existing element could I add between <queue> and <audio sink>,
> > to be able to turn on / turn off data flow to audiosink?? Is there an
> > element like Identity
> > (
> http://www.gstreamer.net/data/doc/gstreamer/0.10.3/gstreamer-plugins/html/gstreamer-plugins-identity.html),
> but with additional feature of stopping the data flow?
>
> You could use identity drop-probability=1.0, or the 'valve' element from
> gst-plugins-bad (to be moved to core, -base or good soon hopefully). You
> would probably also want to set the "async" property of the actual
> audiosink element to false then.
>
> Cheers
> -Tim
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Tue, 19 Oct 2010 11:37:37 -0400
> From: "Gruenke, Matt" <mgruenke at Tycoint.com>
> Subject: Re: [gst-devel] long pauses when viewing RTSP stream
> To: "Discussion of the development of GStreamer"
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <C575E3EFD5C26F46A7CB8A79B9CD399307C93441 at lxi1exc02.americas.tsp.ad
> >
> Content-Type: text/plain; charset="us-ascii"
>
> Is the 241Q running the latest firmware?
>
> If you change the 'latency' property of rtspsrc, does it affect the
> amount of time spent "pausing"?
>
>
> Matt
>
>
> -----Original Message-----
> From: Doug Crawford [mailto:dcraw101 at yahoo.com]
> Sent: Friday, October 15, 2010 15:33
> To: gstreamer-devel at lists.sourceforge.net
> Subject: [gst-devel] long pauses when viewing RTSP stream
>
>
> I am viewing a RTSP stream from an AXIS 241Q video server and displaying
> it
> on my OMAP3EVM board. My gstreamer pipeline is: gst-launch rtspsrc
> location=rtsp://10.5.5.33/mpeg4/media.amp ! decodebin2 ! TIDmaiVideoSink
> videoStd=VGA videoOutput=LCD sync=false rotation=90
>
> The video pauses for about 5 seconds then plays very fast for about 2
> seconds and this keeps repeating over and over. Any ideas?
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/long-pauses-when-viewing-RTS
> P-stream-tp2997576p2997576.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
> ------------------------------------------------------------------------
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> ------------------------------
>
> Message: 5
> Date: Tue, 19 Oct 2010 17:55:07 +0200
> From: Andoni Morales <ylatuya at gmail.com>
> Subject: Re: [gst-devel] DV capture pipeline frozen
> To: Discussion of the development of GStreamer
> <gstreamer-devel at lists.sourceforge.net>
> Message-ID:
> <AANLkTim6z=+N1JhEkLFkHC0CLmFowrG4Tfo24WnxvAf2 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> 2010/8/16 Gregory Petrosyan <gregory.petrosyan at gmail.com>:
> > On Mon, Aug 16, 2010 at 11:35 PM, Gregory Petrosyan
> > <gregory.petrosyan at gmail.com> wrote:
> >> I am having some problems with GStreamer. Basically, I have a
> >> pipeline, which used to work (with Ubuntu 9.04 GStreamer packages).
> >> Here it is:
> >>
> >> ...
> >>
> >> Basically, it captures video from a DV camera, stores raw DV data,
> >> encodes it to H.264 on the fly and shows video preview window.
> >
> > Here are the minimal pipelines, which reproduce the problem:
> >
> > This works:
> > gst-launch-0.10 -e dv1394src ! queue ! dvdemux ! ffdec_dvvideo ! queue
> > ! ffmpegcolorspace ! x264enc ! mpegtsmux ! queue ! filesink
> > location=test.avi
> >
> > And this:
> > gst-launch-0.10 -e dv1394src ! queue ! dvdemux ! ffdec_dvvideo ! tee
> > name=t ! queue ! ffmpegcolorspace ! x264enc ! mpegtsmux ! queue !
> > filesink location=test.avi t. ! queue ! ffmpegcolorspace ! xvimagesink
> > sync=false
>
> The x264 encoder needs some buffers before the pushing first one
> downstream, which full the queue before the video sink.
>
> Disable the limits in the queue (queue max-size-bytes=0
> max-size-buffers=0 max-size-time=0) and that will fix you problem:
> gst-launch-0.10 -e dv1394src ! queue ! dvdemux ! ffdec_dvvideo ! tee
> name=t ! queue ! ffmpegcolorspace ! x264enc ! mpegtsmux ! queue !
> filesink location=test.avi t. ! queue max-size-bytes=0
> max-size-buffers=0 max-size-time=0 ! ffmpegcolorspace ! xvimagesink
> sync=false
>
> Next time you can debug it naming the queues and using GST_DEBUG=*queue*:5:
> queue_dataflow gstqueue.c:930:gst_queue_chain:<sink_queue> received
> buffer 0xb53029f0 of size 153600, time 0:00:01.033333333, duration
> 0:00:00.033333333
> queue_dataflow gstqueue.c:963:gst_queue_chain:<sink_queue> queue is
> full, waiting for free space
> queue_dataflow gstqueue.c:968:gst_queue_chain:<sink_queue>
> (sink_queue:sink) wait for DEL: 30 of 0-200 buffers, 4608000 of
> 0-10485760 bytes, 1000000000 of 0-1000000000 ns, 30 items
>
> As you see the limit in time was reached in the queue "1000000000 of
> 0-1000000000 ns"
>
> Andoni
> >
> > results in frozen preview window + zero-length test.avi file.
> >
> > ? ? ? ? ? ? ? ? Gregory
> >
> >
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>
>
>
> --
> Andoni Morales Alastruey
>
> LongoMatch:The Digital Coach
> http://www.longomatch.ylatuya.es
>
>
>
> ------------------------------
>
> Message: 6
> Date: Tue, 19 Oct 2010 12:46:13 -0700 (PDT)
> From: Wes Miller <wmiller at sdr.com>
> Subject: Re: [gst-devel] Choppy Audio over UDP
> To: gstreamer-devel at lists.sourceforge.net
> Message-ID: <1287517573558-3002698.post at n4.nabble.com>
> Content-Type: text/plain; charset=us-ascii
>
>
> Hi All,
>
> Two additional bit of information:
>
> 1. "TURN THAT $@!# THING DOWN" I added volume elements in both sener and
> receiver pipes. Made a big difference. Guess I was overdriving
> everything.
>
> 2. For the aac pipes above, I had to slow down the clock-rate on the
> receiver to about 20400 to get something that sounded even remotely like it
> was matched to the clock-rate=44100 sender.
>
> So, it's still awfully jittery. On a whim, I tried going back to just
> using
> udpsrc/udpsink without rtpbin. Still poor quality. Then I took out the
> dmaienc_aac and replaced it with several different encoders (aka, whatever
> TI and RidgeRun managed to stick in the GST packages). Finally landed on
> alawenc/dec. Suitably altered the clock-rate and nixed gstrtpbin and
> behold, pretty good sound. A mite echoy but WAY better.
>
> So, these are the best pipes I have right now:
>
> SENDER:
>
> gst-launch-0.10 -e -v \
> alsasrc do-timestamp=true \
> ! queue2 \
> ! alawenc \
> ! udpsink port=5002 host=$1
>
>
> RECEIVER:
>
> gst-launch-0.10 -v \
> udpsrc caps="audio/x-alaw, channels=2, rate=29000" \
> port=5002 \
> ! queue2 \
> ! alawdec \
> ! volume volume=0.1 \
> ! queue2 \
> ! pulsesink
>
>
> Thanks for all the help, M4arco.
>
> Wes
>
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3002698.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
>
>
> ------------------------------
>
> Message: 7
> Date: Tue, 19 Oct 2010 21:51:30 +0200
> From: Arnout Vandecappelle <arnout at mind.be>
> Subject: Re: [gst-devel] Reduce latency for a MJPEG over UDP multicast
> pipe
> To: gstreamer-devel at lists.sourceforge.net
> Cc: STJME <jonas.melin at saabgroup.com>
> Message-ID: <201010192151.31296.arnout at mind.be>
> Content-Type: Text/Plain; charset="us-ascii"
>
>
> On Friday 08 October 2010 15:32:22, STJME wrote:
> > Another thing is that it appears as if the RTP protocol adds substantial
> > delay (70ms or so). By using raw UDP we could decrease the delay.
> However,
> > that is quite uggly. The best thing would be to try to tweek the RTP
> stack.
> > We have tryied, but cannot see any effect. Do you know anything about
> that?
>
> I didn't know about that... At reception side, RTP has a jitterbuffer
> which
> reorders packets and compensates for clock and network jitter, but I guess
> you
> already configured that down to the minimum.
>
> Do you know if this latency is caused by the sender or by the receiver?
>
> Regards,
> Arnout
>
> PS If you want a quick reply, CC me, since I don't read the list very
> often
> :-)
>
> --
> Arnout Vandecappelle arnout at mind be
> Senior Embedded Software Architect +32-16-286540
> Essensium/Mind http://www.mind.be
> G.Geenslaan 9, 3001 Leuven, Belgium BE 872 984 063 RPR
> Leuven
> LinkedIn profile: http://www.linkedin.com/in/arnoutvandecappelle
> GPG fingerprint: 31BB CF53 8660 6F88 345D 54CC A836 5879 20D7 CF43
>
>
>
> ------------------------------
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> End of gstreamer-devel Digest, Vol 53, Issue 52
> ***********************************************
>
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