, they fall back on TCP.
What I wanted to know was if our server could always force the client to
connect in TCP. As far as I know that would imply altering the m= line(s
) in the SDP description, am I correct? Would it be possible to do that
easily in LiveMedia?
Thank you,
Cristiano.
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Il 06/04/2011 18:49, Cristiano Belloni ha scritto:
Il 06/04/2011 12:04, Cristiano Belloni ha scritto:
Hi to all,
I wrote a custom shared memory source. it inherits from FramedSource.
The shared memory is synchronized via Linux semaphores (simple
producer-consumer algorithm), but since I didn&#
Il 06/04/2011 12:04, Cristiano Belloni ha scritto:
Hi to all,
I wrote a custom shared memory source. it inherits from FramedSource.
The shared memory is synchronized via Linux semaphores (simple
producer-consumer algorithm), but since I didn't want to subclass
TaskScheduler, I still
elect() did return but
still there's no data available? I'm getting really confused on this.
Thanks and regards,
Cristiano Belloni.
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r only when you add your own custom
code, I'm not going to be able to help you fix it.
Dear Ross,
LiveJPEGVideoRTPSink adds only a parameters struct to the class members;
it does nothing else. Anyway, I'm gonna remove it and investigate further.
Regards,
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lloc, but the
crash always come in the OutPacketBuffer destructor, presumably after a
free() is made.
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nd use
instead liveMedia's JPEGVideoRTPSink class, the double-free corruption
problem is still here, as it was before commenting out the efficiency
hack. This means that, at least for me, even your post-2010.10.28 code
has problems with the efficiency hack (or so it seems).
Regards,
Cr
the
fOutBuf OutPacketBuffer. When the buffer is deleted, it tries to dealloc
4 gigabytes, and glibc gets very angry.
Solution: commenting out the efficiency hack seems to make the whole
thing work (but the CPU toll without the efficiency hack is too high).
Regards,
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nterval == 0) return 0;
else return 64;
I don't know why you use 1, but in this case you could need to return 65
(64 is the same type of 0, but with DRI markers, and so on).
Also, please don't top-post :)
Cristiano.
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x27;m done with the "double free" issue (I'm investigating!),
I'll gladly post a patch for MJPEG!
Thanks & Regards,
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on */);
Finally, reflect the size change in the headers in specialHeaderSize():
// We assume that there are 'restart markers'
headerSize += 4;
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-40821000 rw-p 4080 00:00 0
40821000-4090 ---p 40821000 00:00 0
be9a8000-be9bd000 rwxp be9a8000 00:00 0 [stack]
If you need more info, I'm gonna collect them as soon as I can.
Best Regards,
Cristiano.
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very old; our RTSP client
implementation has changed significantly since then.
Yes, I can confirm it occurs even with the Oct 20 / 2010 version of
openRTSP. I'm downloading the latest version right now and will make you
know if everything's ok as soon as I test it extensively.
As usual,
IONS
rtsp://192.168.1.97/video RTSP/1.0
CSeq: 1
User-Agent: openRTSP (LIVE555 Streaming Media v2010.04.09)
(second packet, *not* ignored this time).
The problem is found randomly in the systems. Probably it depends on the
net stack, networking hardware and/or network load and lag.
Can you
.
Works like a charm. Thank you.
Cristiano.
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.
I'm getting very tired of this, too, and I can assure you I share your
feelings. I'm sincerely astonished that big manufacturers as Milestone
just can't stick to these simple standards.
I've already contacted Milestone developers and informed them on these
bugs (and on ot
a reference:
http://developer.apple.com/quicktime/icefloe/dispatch028.html , but
unfortunately it doesn't answer my question).
Best Regards,
Cristiano.
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I implement to make RTSP over HTTP work?
Thanks,
Best Regards,
Cristiano.
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ed, because we already use the client's RTCP "RR" packets as a
'liveness' indicator.
(This hack will be removed when we fix the problem with RTP-over-TCP
(RSN, I hope).)
Great! Thank you!
Regards,
Cristiano.
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timeout value? RTSPServer.cpp modify or there's a better solution?
Regards,
Cristiano.
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if the option is listed, VLC has the
right to use it), even if it's a brain-dead schema.
Regards,
Cristiano.
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te VLC, but let every client
download it (its browser-plugin, in fact) straight from the site.
Conversely, I cant' just switch to UDP, as TCP streaming is the only
things that just works in case of blocking firewalls, routers, and all
the sweet things you can encounter on the modern
Cristiano Belloni wrote:
Matt Schuckmann wrote:
We use MJPEG over RTP to send on demand hi resolution still images
along side our video which is lower res H.264.
Is there an extension to the MJPEG standard that allows JPEG over 2k
x 2k? If so we'd be interested in implementing it.
M
to support the extension in liveMedia involves writing support
for RTP header extension in liveMedia lib, and writing an MJPEG source
and sink that can handle the ONVIF headers in your application code
(Ross, correct me if I'm wrong).
Regards,
Cristiano.
Cristiano Belloni wrote:
Ross Finlays
s all
you have, and you got to live with it. That's why people still are still
interested in it (for sake of completeness there's another - secondary -
reason, i.e. retrocompatibility with older software).
Regards,
Cristiano.
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If not, should we develop support for this subset of ONVIF standard,
and the code shoul be ok for you, would you eventually include it in
live555 liveMedia libraries' future releases?
Thanks,
Best Regards,
Cristiano.
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session (aka return a
NULL pointer).
Would it work?
Additionally, the increase and decrease counter operations wouldn't have
to be mutually exclusive because the library works in a non-threaded
fashion, is it right?
Thanks in advance,
Cristiano.
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t
package just this morning. I see only SRs from the server, every 4 sec,
and no RR from the client.
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ct sender
reports, while the client (located at .84) never gives back any RR.
http://img216.imageshack.us/img216/9380/screenshotws.jpg [screenshot]
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liveMedia package (though I can
observe this behaviour in older builds, too) without modifications.
Thanks and best regards,
Cristiano Belloni.
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it works
>> perfectly.
>>
>
> That must be because you are streaming from a live source. If you
> are streaming from a live source, then it's perfectly OK (in fact,
> preferred) to not set "fDurationInMicroseconds" (which means that it
> retains it
ect. (Note the numerous examples of this in the source
> code.)
>
I've always wondered what value "fDurationInMicroseconds" should take in
video frames.
A simple answer is probably 1 sec / framerate. Is it toio simple or am I
correct?
Anyway, I set it to 0 in
e a character
> other than "/" in the RTSP URLs - e.g.,
> rtsp://host/video:MPEG4:stream[number]
> ? This would require no change to the existing code.
>
To achieve URL-uniformity with existing IP cameras, for example. Nearly
all of them use URLs like that.
Cristiano
e
file does not exist in the filesystem).
Regards,
Cristiano.
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to divide it by 1000.
I know it only affects the frequency of RTCP packets, but it's just to
be sure.
Regards,
Cristiano.
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Cristiano Belloni wrote:
> [EMAIL PROTECTED] wrote:
>
>> 1. Are you inheriting from FramedSource, from DeviceSource, or
>> modifying DeviceSource?
>>
> Inheriting from FramedSource.
>
>> 2. Is "isAwaitingData()" function an abbreviation for
27;m experimenting modifying taskScheduler, and I'm rewriting
the code. A soon as I have some readable code, I'll gladly send it to you.
Many thanks,
Regards,
Cristiano
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alled anytime a
DelayQueue::HandleAlarm calls AlarmHandler::HandleTimeout (which in turn
calls handleTimeout, and so on).
Please note I can only see *if* a frame is ready. When / where in the
code should I check that, in order to get my doGetNextFrame called?
Regards,
Cristiano
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frames until the next frame is acquired, filling
rapidly the CPU resources.
Thanks.
Regards,
Cristiano.
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Cristiano Belloni wrote:
> Hi, I'm streaming MPEG4 and use FFMPEG library to encode MPEG4 frames
> from my own live source.
>
> I wrote my own FramedSource subclass and that does the trick.
>
> The problem is, MPEG4ESVideoRTPSink does not set the marker bits at
> all.
nd set the
pictureEndMarker everytime I have a complete frame? Is there no way to
have my marker bits set just subclassing FramedSource, just like I did
with H263 streaming?
Thanks and regards,
Cristiano.
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Cristiano Belloni wrote:
> Ross Finlayson wrote:
>
>> Cristiano,
>>
>> What you're doing looks OK. Perhaps the first thing you could look
>> into is to check whether your RTPSink (subclass)'s destructor is
>> somehow getting called more
nnounceStream(rtspServer, sms, streamName, inputSink);
Am I doing something wrong?
Regards,
Cristiano.
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Ross Finlayson wrote:
> (A reminder: Serious professionals do not use "@gmail.com" email addresses :-)
>
I'm curious about that. Why? :)
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n advance,
Best Regards,
Cristiano.
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rames at once, or to add "out of band" data to a
frame.
Thanks anyway,
Cristiano.
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;s not, is there a way to stuff some special data in the frame
that will be read from my custom sink, so that the sink could tell it's
two frames instead of one and divide them accordingly?
Thank you very much,
Cristiano.
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Cristiano Belloni wrote:
> Because it will mean creating also my own
> subclass of "MediaSubSession", and of "MediaSubsessionIterator", as far
> as I know.
Errata corrige: should be: 'Because it will mean creating also my own
subclass of "MediaSession
ay to go? Or there's another way of using my custom
MultiFramedRTPSource subclass without changing all these other classes?
Thanks,
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my internal buffer,
and I need to copy it at the beginning of the packet (removing the
header in the meanwhile). How can I do this?
Thanks,
Cristiano.
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to set up a function that's called every time an RTP
packet is received?
2) If there is, will the packtes be in order (i.e. function for packet n
is called _after_ function for packet n-1 and _before_ packet n+1)?
3) Is there a way to set up a function that's called every time an RT
>> > I'm saying that you might not need to write any new (sub)classes at
>>> all. Can you tell me some more about what your data looks like -
>>> i.e., what video codec are you using, and how is the data different
>>> from the standard?
>>>
>>It's MPEG-4 data, 1440 bytes a packet.
>
> MPEG-4 f
Cristiano Belloni wrote:
> Ross Finlayson wrote:
>
>>> Ross Finlayson wrote:
>>>
>>>
>>>>> However, I need some info: the camera I'm contacting does not send normal
>>>>> video frames, but custom ones. I need to
, read the
header, accumulate the MPEG-4 data following the header and, as soon as
I have an MPEG-4 frame, pass it to a decoder.
Is there something like a function that is called everytime I get an RTP
packet, so I can examine its payload?
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ssing/decoding of these
> custom frames.
>
>
Ehm, sorry, I didn't understand your answer. Do you mean I should
subclass RTPSource?
What's exactly the object that reads from it?
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Hi, parallel to my RTSP server I'm developing a client to interface with
some ip-cams.
I read openRTSP source, and I modified it to my needs, and it works
wonderfully.
However, I need some info: the camera I'm contacting does not send normal
video frames, but custom ones. I need to unpack every RT
m this, because you're working with your own custom code,
> there's not a whole lot more I can suggest.
>
Do you see any errors in the server code I posted?
Thanks,
Cristiano.
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[EMAIL PROTECTED] &
fifo) and blocked write(s) in the fifo (that
becomes full while the server is streaming).
Maybe it has something to do with the estBitrate in
H263plusVideoFileServerMediaSubsession (too large or too short)? Should
I subclass and change it?
Regards,
Cristiano.
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Errr.. I mean "I managed to figure out".
> At last, I maneged figured out the problem.
>
>
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Cristiano Belloni wrote:
>> You will need to figure our what about your data is causing the
>> parsing to fail. (It is also not inconceivable that our parsing code
>> is in error; if so, let is know.)
>>
>>
>
> Good. I'll go for it and ho
> You will need to figure our what about your data is causing the
> parsing to fail. (It is also not inconceivable that our parsing code
> is in error; if so, let is know.)
>
Good. I'll go for it and hopefully let the list know what went wrong.
Thanks for your help!
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0 seconds to send a 320Kb
raw file from the server to openRTSP (on the loopback interface!)
This is only when i send a pre-recorded raw H263 stream.
Have you idea on what could go wrong?
Thanks.
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> we implement in our code).
This relieves me :) I was starting to fear I would have to write
H263VideoFileServerMediaSubsession.
Thanks.
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leServerMediaSubsession is meant to send H263+
streams, which it does, but I thought H263+ was backwards compatible
with H263.
Anyone managed to stream pure H263 first version with liveMedia?
Anyone knows how can I achieve this goal?
Thanks to everybody.
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ad()s may block
until the thread write()s the freshly encoded frames in the fifo?
Or maybe it is something about concurrent threading? I already tried to
start another process instead of a thread to do the encoding and writing
on the fifo, but I had no luck. same behaviour.
Did it
stream? Now I'm testing if thread is already started, because starting
it in createNewStreamSource would have it started twice. Am I doing the
right thing?
Regards,
Cristiano Belloni.
Cristiano Belloni wrote:
>> I suggest defining a subclass of
>> "H263plusVide
y class, I get similar errors.
Where am I doing wrong? Can you point me to some sample code where it is done
right?
Thanks,
Regards,
Cristiano Belloni.
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>You're going to have to figure out yourself why the server is not
>streaming. (Remember, You Have Complete Source Code (for the
>server), and also a working client.) I'm not surprised, however,
>that RealPlayer doesn't wor; it's notoriously non-standards compliant.
Also, the odd thing is tha
>I presume you're using a "H263plusVideoFileServerMediaSubsession"
>object for your video.
I do.
>Be sure to set the "reuseFirstSource"
>parameter to "True", because you are streaming from a single (live)
>source, rather than from a file that's stored on disk.
I did.
>You're going to have to
an "on demand" fashion.
How can I do that? I thought about starting the decoder thread in
specialClientAccessCheck method, subclassing RTSPServer class, but I
don't think it's the right way to do it (after all,
specialClientAccessCheck is for checking
ynamic, wasn't it ? (tonight I was
reading the code and couldn't explain that sort of behaviour).
Thank you so much,
regards,
Belloni Cristiano
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Ross Finlayson wrote:
>/ char const* inputFileName = "video-H263-1998-1.3pg";
/
> What is the format of this file? Our 'H.263+ parsing' code assumes
> that the input file contains raw H.263+ video elementary stream data
> *only*. If (judging by the ".3gp" filename suffix) your file is
> i
ut I don't know how to avoid it).
I think the correct solution is using the onDemand approach, but does
someone know the reason of its strange behaviour (and how to crcumvent
it, of course)?
Best regards,
Cristiano.
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