Peng,
I see that the frame rate is set to 29.97 ( 6 / 1001 / 2) but I don't see
any other timing information.
Did you get the file to work?
Markus.
seq_parameter_set_data()
profile_idc = 100
constraint_set0_flag = false
constraint_set1_flag = false
co
The file has a number of problems.
Here a log output from a TS checker for file ChunMyung1.ts.
ErrorTotal count Description of last error
PID of last error
Jacob,
You need an H.264 encoder before live555 will stream it.
You could use x264 http://www.videolan.org/developers/x264.html or ffmepg with
x264 or any other H.264 compressor.
You may have to do a little post processing on the compressed stream: adjust
for start codes, AU delimiters and PPS/
How can I fix this?
Thanks
Markus.
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, May 02, 2013 2:49 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] playSIP - creates empty file
I am tryi
I am trying to record the audio from a Polycom Telepresence m100 SIP software
client.
On 10.0.71.24 I run a software VTC client configure to use SIP (Polycom
Telepresence m100).
The URI is sip:10.0.71.24@10.0.71.24
On 10.0.71.109 I run the live555 command line tool playSIP calling 10.0.71.24.
I decode H.264 from a network camera.
My decoder wants SPS/PPS once before the first I-frame in band delivered.
So I wait until I get the first I-frame and then insert
0x001/SPS/0x001/PPS before the I-frame.
Interestingly - one of the cameras I test with sends SPS/PPS inside the RTP
stre
VLC recognizes the correct length but no video and no audio.
I am comparing the live555 saved file with a "clean" mpeg-2 TS file from my
video camera.
On the video camera each I-frame is "prefixed" with AUD, SPS, PPS and the other
picture types (P and B) are prefixed with AUD and PPS.
The PES st
Ross,
> Grumble. What's happening here is not infinite recursion, but 'lots of'
> recursion - caused by the fact that some of your input frames (H.264,
> presumably) are so ridiculously large.
The frames are indent bigger - the camera is HD with a decent bitrate.
I changed the stack size to 8
Ross,
Thanks for your answer.
I tried it and I still get a stack overflow.
1) I had to modify the buffer size in
liveMedia\MPEG2TransportStreamFromESSource.cpp(198) due to the following error
message
MultiFramedRTPSource::doGetNextFrame1():
The total received frame size exceeds the clie
!RecordingClient::Run() Line 187 + 0x33
bytes C++
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Markus Schumann
Sent: Wednesday, April 17, 2013 4:55 PM
To: LIVE555 Streaming Media - development & use
Subject: [Live-devel] saving incoming R
Ross,
Thanks for your initial answer!
I am still trying to save an incoming RTP AVC/AAC stream inside a MPEG-2
transport stream.
I followed the flow of your testRTSPClient.cpp example which worked perfectly
for my first application.
I created a MPEG2TransportStreamFromESSource, FileSink and
I saw various source files (e.g. MPEG2TransportStreamMultiplexor.cpp) muxing
into MPEG-2 TS.
Is there a way a save and/or rebroadcast incoming RTP streams as MPEG-2 TS with
the given code.
Thanks
Markus.
___
live-devel mailing list
live-devel@lists.li
1.)
I compiled and ran live555 on Raspberry Pi but I changed the type of SOCKLEN_T:
Groupsock/include/NetCommon.h
#if defined(__WIN32__) || defined(_WIN32) || defined(_WIN32_WCE)
#ifndef SOCKLEN_T
#define SOCKLEN_T int
#endif
#else
/* Unix */
#ifndef SOCKLEN_T
#define SOCKLEN_T unsigned int
#end
Works like a charm - thanks.
Markus.
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Monday, January 21, 2013 2:22 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] stream descriptor only reachable by
Ross,
Thanks for your input.
I reported the absence of RTSP for the unicast case to the server maker.
Luckily I can configure the device to multicast - here is the SDP
http://10.0.70.25/stream.sdp outputs:
v=0
o=- 8500488 8500488 IN IP4 10.0.70.25
s=ESP H264 STREAM
e=NONE
t=0 0
m=video 8800 RTP
All,
I have an RTP source where it's stream descriptor is only available via HTTP -
any advice on how to go about it?
Thanks
Markus.
URL: http://10.0.70.25/stream.sdp
Browser output:
v=0
o=- 8161451 8161451 IN IP4 10.0.70.25
s=ESP H264 STREAM
e=NONE
t=0 0
m=video 8800 RTP/AVP 96
c=IN IP4 10
Thanks for the response!
I am using live555 as RTSP/RTP client with a custom H.264 and audio sink
connecting to a commercial of the shelve IP camera.
My earlier statement was wrong the timestamps are increasing. The absolute
value of the negative timestamp is going down (see below).
Life is good
All,
I am getting a negative value for presentationTime.tv_sec in
MySink::afterGettingFrame.
I absolute value increases (when ignoring the sign)
In the live555's custom sink sample code the timestamps (of type timeval) are
getting casually casted to an unsigned.
Is it save to ignore the sign
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