Hello,
while streaming MPEG2 or MPEG4, when do out-of-order or too late
arriving frames get discarded?
For example, I guess that an I-frame experiencing a delay of 1 second on
its way will not be of use any more.
I had a look at MultiFramedRTPSource and ReorderingPacketBuffer. Is
ReorderingPacketB
Ross,
It didn't take me days. I have solved my problem of synchronization
that was caused by wrong "presentation time" decoded by the library.
I have had to modify the library as explained in my email exchange.
> At the RTP receiver end, you get - for each frame of media
> data - a presentatio
I agree, it is not really a bug. In fact, your library is used to stream live
DVB-T MPEG2 Transport stream in real-time over the network. So, I think it will
be hard to calculate indexes before streaming the recording TS files.
Concerning the proposed changes, I try to find a better way to corre
Ross,
Give me a few days. I have to double check a few things and I will getting
back to you about that. Then I will provide you with some feedback about the
AAC support in Mpeg2 Transport Stream if it work for me.
Thanks for you great support.
Best Regards
Guy
> -Original Message-
> F
You need to stop thinking about RTP timestamps. Instead, think about
presentation times.
At the RTP sender end, you provide - for each frame of media data - a
presentation time.
At the RTP receiver end, you get - for each frame of media data - a
presentation time. This presentation time is u
fTimestampBase: 0xae0daf0d, tv: 683382.201906, RTP timestamp: 21097
sending REPORT
fTimestampBase: 0xae0daf0d, tv: 683383.099932, RTP timestamp: 101919
Creating RTCP SR packet, SSRC is 0x6da5, NTP is : tv: 683383.099932,
TimeStamp is: 101919
sending RTCP packet
80c80006 6da5 83b4ebf7 1
>I need to stream line-in information.I have converted the PCM
>samples to MP3 ,but how do I change the default "test.mp3 " in
>mp3streamer and what do I write instead of that ?
Please read the FAQ.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_
I need to stream line-in information.I have converted the PCM samples to MP3
,but how do I change the default "test.mp3 " in mp3streamer and what do I
write instead of that ?
thanks
Aravind
___
live-devel mailing list
live-devel@lists.live555.com
http://
fTimestampBase: 0xae0daf0d, tv: 683382.201906, RTP timestamp: 21097
sending REPORT
fTimestampBase: 0xae0daf0d, tv: 683383.099932, RTP timestamp: 101919
Creating RTCP SR packet, SSRC is 0x6da5, NTP is : tv: 683383.099932,
TimeStamp is: 101919
sending RTCP packet
80c80006 6da5 83b4ebf7 199524
>What do you suggest is the best route for determining these settings
>dynamically
I don't know of a good way to do this. As far as I know, everyone
who uses the "wis-streamer" software defines these statically for
their particular target environment (e.g., using -D
directives on the compiler
I believe I might have ran into a design issue for the TimeStamp creation
when MPEG1or2VideoStreamFramer is used. Here what I have found.
I have checked the RTCP SR packets that are sent when I play a video Mpeg2
ES. I have used the sample application testMPEG1or2VideoStreamer. I have
followed th
>I need to transmit data
What kind of data is this?
Well, for now it's not a media file like live555 stream usually, but
a collection of media files (images, informations about those
images, etc.) I receive in a buffer, with packet headers to delimit
them.
I understand live555 is used to st
>hi,
> I am a newer to livemedia. I read the RTSPOverHTTPServer.cpp,
>and found some of the code are masked, does the rtsp-over-http
>function not completed yet?
No, it's still 'work in progress'. Please be patient; it will be
available (and supported in the "LIVE555 Media Server") soon.
-
Hi Ross,
First of all, thank you for your quick responses!
2008/1/31, Ross Finlayson <[EMAIL PROTECTED]>:
>
> >I need to transmit data
>
> What kind of data is this?
Well, for now it's not a media file like live555 stream usually, but a
collection of media files (images, informations about those
hi,
I am a newer to livemedia. I read the RTSPOverHTTPServer.cpp, and found
some of the code are masked, does the rtsp-over-http function not completed
yet? Thanks.
araluni
_
手机也能上 MSN 聊天了,快来试试吧!
http://mobile.msn.com.cn/___
15 matches
Mail list logo