gmane.comp.video.gstreamer.devel

Re: How to save incoming h264 stream without re-encoding?

Mailing List SVR — 2012-05-25T23:10:13

Il 26/05/2012 00:34, W.A. Garrett Weaver ha scritto:

add h264parse after the depayloder will work with all the muxer you 
tryed, add also -e to gst-launch,

Nicola


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Re: How to save incoming h264 stream without re-encoding?

W.A. Garrett Weaver — 2012-05-25T22:34:17

When I tried that, filesink makes a file that is 0 bytes. In theory doing
that should work fine, but when I tried it: it didn't.

That's funny that it works for you, I have no idea why it doesn't work for
me, it would help if rtph264depay was documented. Is it my version of
gstreamer? I'm using 0.10

On Friday, May 25, 2012, Nathanael D. Noblet <nathanael< at >gnat.ca> wrote:
sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\=\",
cameras and only remux to a file)...
doesn't that work for you?

Re: gstreamer on mac

Alexander Horn — 2012-05-25T19:21:51

Artem,

homebrew is still pretty new, not as useful for me yet.
(i tried for a while, but i switched back to macports :)

on macports, just:

sudo port install gst-ffmpeg gst-plugins-bad gst-plugins-base
gst-plugins-gl gst-plugins-gl gst-plugins-gl gst-plugins-good
gst-plugins-ugly gst-rtsp-server gstreamer py27-gst-python

sorry for delay, hope this is useful to you or someone else




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Re: How to save incoming h264 stream without re-encoding?

Nathanael D. Noblet — 2012-05-25T18:45:37

I do something somewhat similar (I capture encoded video from IP network 
cameras and only remux to a file)...

What happens if you have rtph264hdepay ! matroskamux ! filsink??? Why 
doesn't that work for you?

Re: HD video - fast playback problem

Emile Semmes — 2012-05-25T18:26:08

Before I assume too much, ie demuxers, etc, make sure your seek type 
parameter in either your gst_element_seek() or gst_event_seek_new() call 
is GST_SEEK_FLAG_SKIP. Also try or'ing that with GST_SEEK_FLAG_KEY_UNIT 
to seek by keyframes. I haven't tried the KEY_UNIT with a rate change, 
so test that out first. A GST_SEEK_FLAG_FLUSH may help your slower 
computers as well. I typically flush the pipeline on seeks by default.

Your call may look like this:

gst_element_seek( (GstElement *) pipeline, 16.0, GST_FORMAT_TIME, 
GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT | GST_SEEK_FLAG_SKIP, .... );

HTH,
Emile

On 5/25/2012 7:02 AM, cumaniok wrote:

Re: How to run a 2-to-1 element

iron_guitarist1987 — 2012-05-25T18:09:29

I tried the following and got 

gst-launch videotestsrc pattern=snow ! ntoone name=n n.video_sink n.src !
autovideosink --gst-debug-level=3

basesrc gstbasesrc.c:2519:gst_base_src_loop:<videotestsrc0> pausing after
gst_pad_push() = wrong-state



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Re: HD video - fast playback problem

cumaniok — 2012-05-25T14:02:57

pritesh kumar-3 wrote

thanks for the reply,

I'm new by in GStreamer, so maybe I misunderstood your suggestion, can you
be more explicit.
I've tried to solve this by seeking to the next position, but frame needs
time to be updated, so I got the same frame on the screen even if position
has been updated. 
I have troubles only with  HD video size, with small size videos everything
works fine.

regards.


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Memory not released gstrtpbin on pad removed

tanmay.ambre — 2012-05-25T13:17:15

i am trying to determine why the memory consumed doesnt return to pre - pad
added level even after unlinking and removing all elements added from
pipline.

udpsrc --> gstrtpbin
Memory :: 2640K

on pad-added signal for gstrtpbin (autoremove=true)

I add and link the following elements to the bin 

[gstrtpbin(srcPad)(existing)]-->sink
(queue)-->rtpspeexdepay-->speexdec-->directsoundsink

The stream starts playing Memory ::3120K - Everything fine

When pad-removed signal is recieved

I unlink ,remove, Set state to NULL and unref for the following elements
-->sink -->rtpspeexdepay-->speexdec-->directsoundsink
The ref count at the end of the function for each element shows 0.

i expect the memory to got back to its previous level of 2640K But instead
it is 2750K.

Am I missing some obvious part i need to do. Or how can I trace this
increased memory size?

I am using OSSBuild.

Cheers

Tanmay Ambre

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Re: how to optimize loading time of flash video.

Kasthuri Jayachand Yadlapalli — 2012-05-25T10:50:18

Hi ,


 While loading the 200MB flash  video.Gstreamer is taking long
 time.I had small fix to reduce the loading time for flash video.
This fix is done on flvdemux of gst-good plugins 0.10.21 version.

Please find the attachment of patch for fix and give a comment on this.

Thanks and Regards,
Y.K.JAYACHAND.
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Re: How to save incoming h264 stream without re-encoding?

W.A. Garrett Weaver — 2012-05-25T03:54:09

I found the solution to the intermittent delay problem. What you have to do
is reduce the latency on gstrtpjitterbuffer to 10ms instead of being 500ms.

gst-launch-0.10 udpsrc multicast-group=224.1.1.1 auto-multicast=true
port=5010 caps='application/x-rtp, media=(string)video,
clock-rate=(int)90000, encoding-name=(string)H264,
sprop-parameter-sets=(string)\"Z0KAHukBQHpCAAAH0AAB1MAIAA\\=\\=\\,aM48gAA\\=\",
payload=(int)96, ssrc=(uint)3315029550, clock-base=(uint)3926529534,
seqnum-base=(uint)45576' ! gstrtpjitterbuffer drop-on-latency=true
latency=10 ! rtph264depay ! ffdec_h264 ! x264enc ! matroskamux ! filesink
location=movie.mkv

The result is this:
http://www.youtube.com/watch?v=br--9h3-g4U

This works good if the computer on the receiving end is sufficiently fast
enough to compress video, since mine is, this is an adequate solution.
Still, it would be nice to record the h264 stream directly to a file
instead of having to re compress.

On Wed, May 16, 2012 at 2:01 PM, W.A. Garrett Weaver <
weaverg< at >email.ar

can't control valve drop property

jomifo — 2012-05-24T19:41:16

I am mixing a video stream with a static image.  That's working fine.  I
would like to dynamically turn the image off/on at certain times.  My idea
was to put a valve after the image stream and use a controller to toggle the
drop property.  But alas, I get an error when trying to create that
controller,

In [45]: gst.Controller( gst.element_factory_make( 'valve' ), 'drop' )

** (ipython:2530): CRITICAL **: gst_controlled_property_new: assertion
`(pspec->flags & (G_PARAM_WRITABLE | GST_PARAM_CONTROLLABLE |
G_PARAM_CONSTRUCT_ONLY)) == (G_PARAM_WRITABLE | GST_PARAM_CONTROLLABLE)'
failed
---------------------------------------------------------------------------
RuntimeError                              Traceback (most recent call last)

/mnt/transfer/<ipython console> in <module>()

RuntimeError: could not create GstController object

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Re: How to run a 2-to-1 element

iron_guitarist1987 — 2012-05-24T15:22:31

Is the code even correct? It compiles and installs, but I haven't been able
to test it (don't know how).

Anyone? 

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Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T14:51:45

Hi Wim,

Thank you for the speedy response. Latest git does indeed seem to fix
this particular issue.

Now the server pipelines seems to play correctly. Next, I need to
configure and verify the clients.

/Martin

On Thu, May 24, 2012 at 3:42 PM, Martin Lund <martin.lund< at >ixonos.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:42:11

Hi,

I'll have to take your word for it since I'm not familiar with the
gstreamer internals.

I'll try to build and test latest from git to see how that works out.

For now, is there possibly a way to bypass this bug in my gst-launch command?

The bug thread you are referring to talks about bypass by forcing
capsfilters but I assume this is meant for the client side so I'm not
sure it applies to my server side of things?

/Martin

On Thu, May 24, 2012 at 3:14 PM, Wim Taymans <wim.taymans< at >gmail.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:16:14

I'm sorry about the formatting of my previous mail. Here is my
gst-launch command in hopefully more readable layout:

gst-launch --gst-debug=4 --verbose --gst-debug-no-color \
    gstrtpbin name=rtpbin \
        filesrc location=battleship.mp4 ! qtdemux name=demux \
            demux.video_00 ! queue ! rtph264pay ! rtpbin.send_rtp_sink_0 \
                  rtpbin.send_rtp_src_0 ! udpsink port=5000 \
                  rtpbin.send_rtcp_src_0 ! udpsink port=5001
sync=false async=false \
                  udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
            demux.audio_00 ! queue ! rtpmp4apay ! rtpbin.send_rtp_sink_1 \
                  rtpbin.send_rtp_src_1 ! udpsink port=5002 \
                  rtpbin.send_rtcp_src_1 ! udpsink port=5003
sync=false async=false \
                  udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1

On Thu, May 24, 2012 at 3:02 PM, Martin Lund <martin.lund< at >ixonos.com> wrote:

Re: Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronizationusing gstrtpbin, unable to link in video_00? (bug?)

Wim Taymans — 2012-05-24T13:14:17

It sounds like https://bugzilla.gnome.org/show_bug.cgi?id=672019 that 
was just fixed.

Wim

Streaming 1080p24 H264/AVC and MPEG4/AAC, lip synchronization usinggstrtpbin, unable to link in video_00? (bug?)

Martin Lund — 2012-05-24T13:02:41

Hi,

I'm on a mission to have a server stream a file containing H264/AVC
and MPEG4/AAC to two other hosts. One of these hosts will play the
video stream and the other will play the audio stream. I need A/V lip
syncrhonization between these two hosts so as far as I understand I
need to use the gstrtpbin plugin in order to utilize the RTP/RTCP
protocol to obtain this kind of synchronization.

I'm playing around with gst-launch to test the RTP/RTCP stuff. My
server pipeline is based on the RTP reference example found here:
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh

This is the basic server pipeline I'm trying to configure (assumes it
all running on the same server, ie. no "host=" stuff):

gst-launch
                \
    gstrtpbin name=rtpbin
                \
        filesrc location=battleship.mp4 ! qtdemux name=demux
                \
            demux.video_00 ! queue ! rtp

Re: send buffer through udpsink

enricom — 2012-05-24T09:56:10

Hello, to do so first of all you need to create the buffer, for instance
with: buffer  =  gst_buffer_new_and_alloc (sizeof(your_data)): this way you
create a buffer with required size. Then you have to put your data into it
by using memcpy(GST_BUFFER_DATA(buffer) , your_data, sizeof(your_data)). 
After this step you need to inject it into your pipeline by using an appsrc
element, with gst_app_src_push_buffer(GST_APP_SRC(appsrc), buffer)). Of
course appsrc needs to be connected to the udpsink element. On the other
side you can use udpsrc, then an identity element. This one, by means of the
handoff signal, tells you when a buffer has been received and so you can
extract data from it.





dennis wrote


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ffenc_msmpeg4v2: failed to encode buffer

padam — 2012-05-24T06:49:27

Hi, reffering to 
http://gstreamer-devel.966125.n4.nabble.com/Gnonlin-question-extract-audio-video-clip-td2305307.html#a3063249
THIS  topic i created this pipe:

gst-launch gnlfilesource name=video
caps="video/x-raw-yuv,width=320,height=240" location=file:///D:\\source.avi
start=0 duration=23000000000 media-start=1188000000
media-duration=23000000000 ! identity single-segment=true ! progressreport
update-freq=1 ! ffmpegcolorspace ! ffenc_msmpeg4v2 bitrate=350000 ! avimux
name=mux ! filesink location=D:\\destination.avi gnlfilesource name=audio
caps="audio/x-raw-int,rate=16000,channels=1" location=file:///D:\\source.avi
start=0 duration=23000000000 media-start=1188000000
media-duration=23000000000 ! identity single-segment=true ! audioconvert !
lamemp3enc bitrate=32 ! mux.

When i launch this sometimes, on the same file, i get that errors:

0:00:11.781250000  2512   00B5F320 ERROR                 ffmpeg .:0:: Error,
Inv
alid timestamp=0, last=0
0:00:11.781250000  2512   00B5F320 ERROR                 ffmpeg
g

Re: How to list elements used by playbin2?

Kyrylo V Polezhaiev — 2012-05-23T20:59:15

Thank you very much! :)

Re: How to list elements used by playbin2?

Mart Raudsepp — 2012-05-23T20:14:27

You could look at a debug log with GST_DEBUG=GST_ELEMENT_FACTORY:3 in
your enviroment to see which elements are created.
However, you need to keep in mind that playbin2 does auto-plugging of
elements based on the media container, used video/audio codec and so on,
so by including elements purely based on such metrics, only the same
kind of media files will be possible to be played then. This may be fine
if your application only uses GStreamer to play also included known
media, not user chosen. Otherwise you should probably take a different
approach in choosing which element plugins to include.