On 7/9/2012 2:15 PM, Michael L. Boom wrote:
I need the unmodified version to compare.
Googling the file name in quotes gave me this: http://goo.gl/SdU3e
I also direct you to the FAQ on modifying the code:
http://www.live555.com/liveMedia/faq.html#modifying-and-extending
*Always* make a pri
I made some change to the code but I have no idea what anymore. I need the
unmodified version to compare. Does anyone have this version? I realize old
code isn't supported or made available but I was hoping someone may have this.
It would save me a lot of time.
#define LIVEMEDIA_LIBRARY_VE
> ...connection pending (10035)
This appears to be the key to your problem. Something has failed - most likely
at the server end, given that you had already stopped receiving RTP packets -
and the client's OS is having trouble reconnecting to the server. (This is
happening below the level of
I am using a derivative of the testRTSPClient in an application. I am
detecting stream failure by looking for missing frames. When I detect
that the stream has been "silent" for too long, I need to close that
stream and reopen it. I know the stream source is still working because
other clients
> What is involved with creating a file source for GSM files?
I don't know what GSM files look like. But if they contain fixed-size frames,
then it should be quite simple to write a "GSMAudioFileSource" class.
I *don't* suggest using "MP3FileSource" as a model; that code is old and crufty
(plu
Hi everyone,
What is involved with creating a file source for GSM files? It appears
that there isn't one at this stage. I took "Mp3FileSource" as an
example, left only doGetNextFrame, MIMEType and getAttributes in the
file. Will I need anything else? Are presentation times important, or
don't
> 1) When I am running OpenRTSP to play H.264 RTSP stream coming from a
> camera, it creates video-H264-1 file. What type of file is that?
It's a H.264 video Elementary Stream file (with MPEG 'start codes' 0x0001
inserted at the front of each NAL unit).
> I tried to run this file in vlc p
Your message (plus the fact that your email address tells me that you're just a
casual hobbyist) suggests that you might not understand how the 'event trigger'
mechanism works. An 'event trigger' is intended to be used by an external
thread to signal an event to your event loop. But I never sa
> My application shows following messages on screen.
>
>
> MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
> exceeds the client's buffer size (40).
This error message means exactly what it says: The size of the buffer that you
provided to the "getNextFrame()" call is too
> So in other words, what is needed is this:
>
> 1) Desktop / Notebook with video cam captures H.264 ---> 2) Server needs to
> receive MP4 over RTP (not LIVE555, this existing server cannot be swapped
> out) ---> 3) Desktop / Notebook app consumes video in format other than
> H.264, this app c
Ross,
Thanks for the response. Unfortunately, the environment in which the needed
solution needs to be deployed is restricted by a few non-negotiables -- such as
the need to store / transcode at the server, and a second client app which
cannot be disrupted. So in other words, what is needed is
> I have a query regarding WAV support in RTSP media Server.
>
> We are planning to implement a rtp client for WAV(PCM) audio file.
Note that both "VLC" and "QuickTime Player" already support this. So you may
not need to write your own client application.
> I want to know the RFC number that
Hey everyone,
I have an IP Camera that provide RTSP streams. I got two streams with names
audio-G726-16-2 and video-MP4V-ES-1. I tried to multiplex these to streams
using the following command of ffmpeg.
ffmpeg -i audio-G726-16-2 -i video-MP4V-ES-1 -acodec copy -vcodec copy
output.avi
It giv
Hi,
I have a query regarding WAV support in RTSP media Server.
We are planning to implement a rtp client for WAV(PCM) audio file.
I want to know the RFC number that the WAV implementation complies to.
I tried searching but could not find the RFC number for the same.
Can someone give any inform
> What is the recommended approach / LIVE555 sample app I should be looking app
> to push MP4/H.264 video to a server via RTP?
The best approach is to have the server directly integrated with your H.264
video source (rather than have the H.264 video source 'pushing' to anything).
I.e., don't s
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