Hi Ross,
I have call every framed source's stopgettingframes, then call
deleteServerMediaSession, why the "aftergettingframe1" is still be called, and
the program would crash at if(!SendRTPorRTCPOverTCP)?
Is there any method to stop the serversession before delete them?
But in the second case, even if I din't parse the data, and every nalu is
started with start code, but live555 worked correctly.
The h264 data client recieved doesn't have start code. So live555 has removed
them?
At 2014-05-29 09:42:15, "Ross Finlayson" wrote:
In each case - because your
Hi Ross,
My system has two type rtsp stream.
One is living video, other device transfer their h264 data to my system, I
copied them to one queue, and the framed source would get one sample every
time, these sample is h264 nalu, and isn't add start code
(0x00,0x00,0x00,0x01).
Other is
Hi Ross,
I have some questions needs your help.
1. I found that when I call removeServerMediaSession, only the audio
and video OnDemandMediaSubsessions are destroyed.
The audio framed source ,video framed source , simple rtp sink, h264
video rtp sink are not destroyed.
Shoul
Oh, I downloaded the latest version. It seems more stable and faster.
THANK YOU FOR YOUR GREAT WORK!!
At 2014-03-31 04:46:58,"Ross Finlayson" wrote:
Are you using the latest version of the "LIVE555 Streaming Media" software??
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
I found another problem,
The seek only can work in one play request.
for example, if open it by vlc and seek it at the first time, then it work.
If I stop this stream, then open it again and seek, it would fail. Is there
any settings to reset when new play request is comming?
At 2014-02
Hi Ross,
I read the faq in live555 website about the trick play, it only said the four
virtual functions needs to be supported.
But as the result, the vlc would drop audio buffer, so I guess the
fPresentationTime should be changed too?
Is it right?_
Hi Ross,
Does live555MediaServer have plan to support mp4 file?___
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I checked the code about TaskScheduler and Environment, RtspServer, it seems
that no interface is stop the rtsp server or exit the event loop when my
application exits.
Should I force it to exit?___
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but unfortunately, vlc also shows that the g711 audio is late.
At 2013-12-24 08:15:00,"Ross Finlayson" wrote:
I saw the DeviceSource code in live555 project.
I found that the the fPresentationTime would be set by gettimeofday when
doGetNextFrame is called.
I remembered you said the fPresen
Hi Ross,
I checked the rtcp live555 sent to client, it only contains SSRC, no rtp and
npt information.
Does I need to create RTCP instance apparently?___
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HI, Ross
Here the code of compute next fPresentationTime.
// Note that the presentation time for the next NAL unit will be different:
struct timeval& nextPT = usingSource()->fNextPresentationTime; //
alias
nextPT = usingSource()->fPresentationTime;
doubl
Hi, Ross,
I saw the DeviceSource code in live555 project.
I found that the the fPresentationTime would be set by gettimeofday when
doGetNextFrame is called.
I remembered you said the fPresentationTime must be same at the first time,
So how should it sync audio and video?
HI,
I need to put two video streams and one audio stream in one rtsp session.
I add the stream by creating subsession, but I found that the two video stream
can't play correctly, sometimes one stream would stop or two streams stopped
too.
Is there any setting to set it ?_
If I did as you said. VLC always said the input buffer is empty.
At 2013-10-26 12:05:12,"Ross Finlayson" wrote:
nextTask() =
envir().taskScheduler().scheduleDelayedTask(2,(TaskFunc*)FramedSource::afterGetting,
this);
[...]
nextTask() =
envir().taskScheduler().scheduleDelayed
But if I use the vlc to play it, the audio's pts is out of range, and it's
dropped.
but it works by ffplay.
I don't know where is wrong.
Could you help me to check it where is wrong. I'm crazy caused by this problem.
Here is my code.
audio is g711, each buffer is 160 bytes.
void AudioFrameSource::
I saw the FAQ in live555 website.
It said the live555 sync the audio and video by RTCP's SR packets.
So I should create RTCP instance for each RTP source explititly?
At 2013-10-24 05:48:40,"Ross Finlayson" wrote:
So how should I set the duration, then the audio and video would be sync.
Yo
Thanks your answer.
There is another question about the scheduleDelayedTask(duration,x,x).
So how should I set the duration, then the audio and video would be sync.
Currently my every audio's frame is 2ms. the video frame rate is 25fps..
Now I set the audio's next getframe time is 2ms, vid
> You can also download and stream this Transport Stream file:
> http://www.live555.com/liveMedia/public/h264-in-mp2t/bipbop-gear1-all.ts
That did the trick, thank you. I knew I had it set up right. That's a
perfect file for my purposes, btw.
Cheers,
t.
On Thu, Mar 14, 2013 at 8:40 PM, Ross Fin
I am using LIVE555 Media Server version 0.76 (LIVE555 Streaming Media
library version 2013.01.03) on CentOS 6.3. It installed and runs fine,
but I am having a hard time finding test media to verify rtsp. I've
tried a variety of mpg files, always some kind of problem with it
(like "StreamParser::par
Thank you for your suggestions. I've downloaded the latest source
code and rebuilt it. It is now:
LIVE555 Streaming Media library version 2013.01.03
welcome.wav still doesn't stream, and since file shows:
# file welcome.wav
welcome.wav: RIFF (little-endian) data, WAVE audio, Microsoft
PCM, 16 bi
Hi,
I have read many thread on the dreaded message "RTSP/1.0 404 File Not
Found, Or In Incorrect Format" when trying to stream from
Live555MediaServer, yet I was not able to find any solution. Below I
attach output from openRTSP client, strace from live555MediaServer,
directory structure, and live
real issue is or just check for a
null value returned from ctime().
Cheers,
Tony.
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to do this it is needed to
change the library to support also stream parser by hardware blocks with
dedicated interfaces.
Since you are already using the library have you got ideas of how to do
that?
Regards,
Tony
___
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live-
zation with parameters x, y, z).
Just to have a tought, since you know the livemedia library better then
me, how many hours/days/month would you think is needed to do the
changes?
Regards,
Tony
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ht
ftware RTSP client / Hardware RTP receiver:
Similar as the server but then the host runs only a RTSP client and the
hardware can receive multiple streams at high bitrates.
Regards,
Tony Vankrunkelsven
Design Engineer
* +32 14 25 27 44
Fax +32 14 25 25 70
* [EMAIL PROTECTED]
Nokia Siemens N
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