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safa vakili | 9 Feb 21:21
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bitrate

Hi everybody,

I am streaming h264 video over the Lan and I want to measure the outgoing bitrate at server.
Would you please help me in this regard?

Thanks,
Safa

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Luca Della Ghezza | 9 Feb 21:16
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how to stream a microphone over a lan?

Well, in a another question in this mailing-list I learned that gstreamer can manage gconf data.
I have a simple script that connect two pc through ssh and open an audio connection using "arecord | aplay" command. Now I wish to switch to gstreamer, because it can use gconf data and detect automatically the audio device for the mic (gconfaudiosrc).
I tried this simple pipe in local and it works:
gst-launch gconfaudiosrc ! volume volume=1.5 ! alsasink

Now I wish to try to stream the mic, this way:
server side (192.168.0.2)
        gst-launch gconfaudiosrc ! tcpserversink host=192.168.0.1 port=5000
client side (192.168.0.1)
        gst-launch tcpclientsrc host=192.168.0.2 port=5000 ! autoaudiosink

(I need to use TCP because the router filters UDP)

But it doesn't work, on the client side it says that there is a format error because it can't negotiate the format.

I hardly tried to search the web looking for a solution, but the few solutions I founded was not working.
Please can someone tell me how I can write the 2 pipes or where I can find documentation about this?

Thank you so much.

Luquino 
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Damián Nohales | 9 Feb 18:43
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PlayBin2: Setting source without "source-setup"

Is there a way to setting up the PlayBin2 source (proxy, user-agent,
etc) without using "source-setup" signal?

This is because I need to support GStreamer since 0.10.20 and
"source-setup" signal is supported since 0.10.33.

Thanks!
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O'neill, Gerry | 9 Feb 16:31
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x264enc causes video to freeze

Hi

I am having problems getting this pipeline to work.

 

gst-launch -v filesrc location=6_avc.mp4 ! \

                qtdemux  ! queue ! ffdec_h264 ! videoscale ! \

                capsfilter caps=video/x-raw-yuv,width=640,height=480 ! \

ffmpegcolorspace ! tee name=myT ! xvimagesink  \

                myT. ! x264enc ! mpegtsmux ! filesink location=surf.ts

 

The problem seems to be with x264enc (works with ffenc_mpeg4).

Any suggestions greatly appreciated?

 

Gman

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Matthias Dodt | 9 Feb 15:52

Avi file duration: gstutils.c: line 586: assertion `denom > 0' failed (OSSBuild) - avidemux problem??

Hi!

I have an avi file containing DV encoded video (PCM Audio). I want to
query the duration of the video, the simple pipeline is (i am using
OSSBuild):

filesrc location=test.avi ! decodebin2 name=decoder

I set the pipeline to pause and execute:

Gboolean res=gst_element_query_duration (pipeline, &time_format, &_dur);

However this always returns false! If i try to access the property via
gst_query_parse_segment, i get the duration in BYTE format. However i
cannot convert it to TIME format:

GstFormat fmt=GST_FORMAT_TIME;
gint64 start=-1,stop=-1,val;

GstQuery *query = gst_query_new_segment (fmt);
gst_element_query (pipeline, query);
gst_query_parse_segment(query,NULL,&fmt,&start,&stop);

duration=stop-start;

if (fmt!=GST_FORMAT_TIME){
	GstFormat oFmt=fmt;
	GstQuery *query=gst_query_new_convert(fmt,stop,GST_FORMAT_TIME);

	//This will cause: gstutils.c: line 586: assertion `denom > 0'
failed !!!!
      if (gst_element_query (g.pipeline(), query)){
		...
	}

However i traced it down to avidemux - where in
gst_avi_demux_src_convert 
it calls
avi_stream_convert_frames_to_time_unchecked (stream, src_value);

Where stream seems to be NULL!

Why?! Am i missing something?

Thanks!

mat
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Farkas Levente | 9 Feb 14:02

rtspsrc enhancement with events

hi,
currently the rtspsrc in gstreamer are not able to handle any kind of
play related events except seek. even seek was implemented at the rtsp
level (send a pause then seek and play). all other events are dropped in
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtsp/gstrtspsrc.c
in gst_rtspsrc_handle_src_event then in
gst_rtspsrc_handle_internal_src_event function which simple drop these
events.
we'd like to implement at least SEEK, RATE, STEP and NAVIGATION event in
rtspsrc to be able to send these events through rtspsrc and the server
get it. so we'd like to play with different rate, backward and frame by
frame through rtsp. and if the server implements it than it can handle
it. (of course we'd like to implement the server side too).

of course we'd like to add the patches to gstreamer, but before we start
to implement it we'd like to ask core gstreamer developers which would
be the preferred way you ie. how to implement it to be easily accepted
by you to inclusion. we thought the best would be to send these events
through the rtcp protocol which is something for it.
we'd like to do it first in the 0.10 branch.

what do you think about it?

thanks in advance.
regards.

ps. i also write a bz about it:
https://bugzilla.gnome.org/show_bug.cgi?id=669746

--

-- 
  Levente                               "Si vis pacem para bellum!"
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Durrand, Paul (Contractor | 9 Feb 04:55
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Multiple appsrc to a single muxer [SEC=UNCLASSIFIED]

UNCLASSIFIED

I am trying to modify the mpegtsmux plugin to process KLV metadata. I am currently having trouble creating a pipeline which will allow me to write a video and metadata with multiple appsrc plugins.
 
A new cap has been created on mpegtsmux named "application/x-klv". I am trying to attach an appsrc element directly to this which will mux with an mpeg2 video via ffenc_mpeg2video.
 
My pipeline looks like this and is called using gst_parse_bin_from_description() :
 
mpegtsmux name="muxer" ! filesink location=gstreamer.out.mpg appsrc name="adsssrc" caps="video/x-raw-rgb,width=704,height=480,framerate=(fraction)25/1,bpp=24,depth=24,red_mask=255,green_mask=65280,blue_mask=16711680,endianness=4321" ! ffmpegcolorspace ! videorate ! ffenc_mpeg2video bitrate=15000000 ! muxer. appsrc name="metasrc" caps="application/x-klv" ! muxer.
 
The mpegtsmux_create_stream() function is never hit when using this pipeline and a video is never created. In the debug there are lots of references to queuing the buffer in gst_app_src_push_buffer_full().
 
If I remove the additional appsrc as below, the megtsmux_create_stream() function is hit (shown through debug):
 
mpegtsmux name="muxer" ! filesink location=gstreamer.out.mpg appsrc name="adsssrc" caps="video/x-raw-rgb,width=704,height=480,framerate=(fraction)25/1,bpp=24,depth=24,red_mask=255,green_mask=65280,blue_mask=16711680,endianness=4321" ! ffmpegcolorspace ! videorate ! ffenc_mpeg2video bitrate=15000000 ! muxer.
 
Can anyone tell me if the syntax for my pipeline is correct for using multiple appsrc with mpegtsmux?
 
Any help would be appreciated.
 
Thanks,
Paul

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max.s | 8 Feb 23:52
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Stream the moviesound, playbin2

Hello gstreamer developers! Im curious If it is possible To stream the sound from the movies i play using playbin2 in my Python script? I should mention that i play the sound in My speakers on the computer already and want to keep that audiosink. Just add one more who stream it to localhost and can I play that sound in a html5 player.. If anyone has any knowledge about this, or if it take a lot of memory on the cpu. I would be very happy if you told me. Regards Max

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Jacques Bride | 8 Feb 18:27

Best way to add a filter inside PlayBin2

Hello

I am developing a frame enhancement filter for the need of a video
control application. I am developing on a OMAP4 pandabaord on ubuntu
with the Texas Instruments hardware accelerated gstreamer pipeline
(1080PPfps decode), anyhow I think my question is rather general.  I
am a beginner in gstreamer so please do not hesitate to redirect me to
the doc if my question is already answered somewhere.

My project consists in a video filter that transforms the yuv decoded
frames before being displayed (for example a sepia filter to display
frames in sepia color).

What is the best way to add my filter in the installed gstreamer
pipeline such that it becomes part of my video chain?

Please see below more details:
- I wish to use the installed and omap4 optimized gstreamer/totem
pipeline to enable full video stream
decode and display with my filter on.
- I created  a gstreamer plugin/element for my filter with the
gst-template framework, but how to make it part of GstPlayBin2?
- I looked at the pipeline graph of totem and gst-launch: they both
use the playbin2 module, and I would like to add my filter in-between
GstURIDecodeBin and GstPlaySink (input from src0 and output to
video_raw_sink  through gstinputselector). Is is possible to add and
link a new element  here without breaking the full bin? It is not
either on the input or the output of playbin2 so I do not really
understand how to just create a playbin2 element in my application and
delicately insert a new element inside...
- do I have to modify the source code of PlayBin2, add my filter,
recompile and install it on my board to have my added features
naturally enabled when I play videos? Or is it possible to tell
existing gst-launch and totem applications to add a user filter plugin
in the installed video processing?

Thanks for your help
Jacques
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Dinesh Iyer | 8 Feb 18:09
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Reading one channel AAC file using playbin2 returns stereo data

Hello,
I am using playbin2 to read audio data from AAC and MP3 files. I noticed
that when I attempt to read one channel AAC files, the output returned is
stereo with both channels having the same data. Single channel MP3 and WAV
files were read in as expected. I am attaching an image of the pipeline
obtained when using the following gst-launch command:

gst-launch-0.10 playbin2
uri=file:///local/playbin2_appsink_aac_mono/testWelcomeMono.m4a
audio-sink=appsink video-sink=fakesink text-sink=fakesink

Regards,
Dinesh

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Dinesh Iyer | 8 Feb 18:09
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Reading one channel AAC file using playbin2 returns stereo data

Hello,
I am using playbin2 to read audio data from AAC and MP3 files. I noticed
that when I attempt to read one channel AAC files, the output returned is
stereo with both channels having the same data. Single channel MP3 and WAV
files were read in as expected. I am attaching an image of the pipeline
obtained when using the following gst-launch command:

gst-launch-0.10 playbin2
uri=file:///local/playbin2_appsink_aac_mono/testWelcomeMono.m4a
audio-sink=appsink video-sink=fakesink text-sink=fakesink

Regards,
Dinesh

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Gmane