[Live-devel] live-devel Digest, Vol 57, Issue 13
soumya patra
soumya.patra at lge.com
Fri Jul 11 20:57:58 PDT 2008
Hi Ross,
Thanks for the concern, now I am able to stream h.264 with correct SDP.
I've corrected the SDP. Now it's streaming.
Regards,
Soumya
LGSI
-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of
live-devel-request at ns.live555.com
Sent: Saturday, July 12, 2008 4:25 AM
To: live-devel at ns.live555.com
Subject: live-devel Digest, Vol 57, Issue 13
Send live-devel mailing list submissions to
live-devel at lists.live555.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.live555.com/mailman/listinfo/live-devel
or, via email, send a message with subject or body 'help' to
live-devel-request at lists.live555.com
You can reach the person managing the list at
live-devel-owner at lists.live555.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of live-devel digest..."
Today's Topics:
1. Re: H264 SDP (Ross Finlayson)
2. live555 Test Programs (Hannah)
3. Re: H264 SDP (Jerry Johns)
4. Re: live555 Test Programs (Ross Finlayson)
5. Re: H264 SDP (Ross Finlayson)
6. Creating a RTSP stream from DVB-T hardware card (Ryan Walklin)
----------------------------------------------------------------------
Message: 1
Date: Thu, 10 Jul 2008 23:12:53 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] H264 SDP
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240802c49ca8b0cfe9@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii"; Format="flowed"
>I have attached our sdp and NAL frame (contains NAL units 7 and 8)
>alongwith the mail. We may be missing some needed information in sdp
>or we might be communicating it in the wrong format or I might be
>totally mistakenJ. Could you tell us the correct sdp fmtp line to
>generate
No I can't, because only you know the details of your H.264 stream.
Are you sure you are encoding the sps and pps NAL units correctly
using Base64 (for the "sprop_parameter_sets_str" parameter)? I
suggest that you go through VLC's LIVE555 interface code (in
"modules/demux/live555.cpp") to make sure that the SDP fmtp string is
decoded into the correct NAL unit data (see the function
"parseH264ConfigStr()").
If you're still having problems, then you may need to make your RTSP
stream publically available, and post to the VLC mailing list, asking
them why VLC cannot play it.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080710/52c0de0b
/attachment-0001.html>
------------------------------
Message: 2
Date: Fri, 11 Jul 2008 20:23:38 +0800
From: Hannah <havthanh at gmail.com>
Subject: [Live-devel] live555 Test Programs
To: live-devel at ns.live555.com
Message-ID:
<c27d1fdb0807110523j47711ab1j4f9ae3dad5eb3604 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
Could I ask you this question?
I compiled the source code already and now testing the test programs. I try
the MP3 Audio test programs as first example.
+ For the testMP3Streamer, the output receiver is "Unable to open file
"test.mp3" as a MP3 file source"
+ For testMP3Receiver, output is "Beginning receiving multicast stream..."
The other test programs have the same output, I can't stream or play any
file. What's the problem and how to solve?
*Secondly*, I run the testOnDemandRTSPServer, and it returns me the URL
links of the audio/video test files, for example :
"mpeg4ESVideoTest" stream, from the file "test.m4e"
Play this stream using the URL "rtsp://10.10.10.103:8554/mpeg4ESVideoTest"
"mpeg1or2AudioVideoTest" stream, from the file "test.mpg"
Play this stream using the URL "rtsp://
10.10.10.103:8554/mpeg1or2AudioVideoTest"
Is it the URL link to use with openRTSP client, that I ask below?
*Thirdly*, I run the RTSP client, by the command : *./openRTSP,* and I see
the usage instructions :
Usage : ./openRTSP [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V] [-d
<duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>] [-n] [-O]
[-t|-T <http-port>] [-u <username> <password>] [-s <initial-seek-time>] [-z
<scale>] [-w <width> -h <height>] [-f <frames-per-second>] [-y] [-H] [-Q
[<measurement-interval>]] [-F <filename-prefix>] [-b
<file-sink-buffer-size>] [-B <input-socket-buffer-size>] [-I
<input-interface-ip-address>] [-m] <url> *(or ./openRTSP -o [-V] <url>)
* I have tried to use the bold part, for instance *./openRTSP -o [-V]
rtsp://10.10.10.103:8554/mpeg1or2AudioVideoTest
*But I received nothing. The terminal screen return the usage instructions
again.
Did I do wrongly? Or do I misunderstand any part?
Is there any solutions or how to solve the situation?
Thanks a lot for your help on this.
Hannah
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080711/e3274f5f
/attachment-0001.html>
------------------------------
Message: 3
Date: Fri, 11 Jul 2008 06:44:31 -0700
From: "Jerry Johns" <Jerry.Johns at nuvation.com>
Subject: Re: [Live-devel] H264 SDP
To: <live-devel at ns.live555.com>
Message-ID:
<274F7B50569A1B4C9D7BCAB17A9C7BE104AB17 at mailguy3.skynet.nuvation.com>
Content-Type: text/plain; charset="us-ascii"
Are you sure you're creating your SDP properly? We've managed to get our
streams working with VLC/Quicktime using proper SDP params
Is your profile-level-id correct? Check the RFC spec on exact details
As for Base64, try using this site to check your vals:
http://www.paulschou.com/tools/xlate/
is your format type (97) correct?
We use 96
Hope it helps,
Jerry Johns
Design Engineer
Nuvation Research Corp - Canada
Tel: (519) 746-2304 ext. 225
www.nuvation.com <http://www.nuvation.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080711/0197ef2f
/attachment-0001.html>
------------------------------
Message: 4
Date: Fri, 11 Jul 2008 07:12:17 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] live555 Test Programs
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240800c49d19636f5a@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii"; Format="flowed"
>+ For the testMP3Streamer, the output receiver is "Unable to open
>file "test.mp3" as a MP3 file source"
Do you have a MP3 file named "test.mp3" in the same directory?
>+ For testMP3Receiver, output is "Beginning receiving multicast stream..."
Note that "testMP3Receiver" outputs its received (MP3) data to
'stdout'. (If you don't understand what 'stdout' means, then this
software is probably not for you.)
>Secondly, I run the testOnDemandRTSPServer, and it returns me the
>URL links of the audio/video test files, for example :
>
>"mpeg4ESVideoTest" stream, from the file "test.m4e"
>Play this stream using the URL
>"rtsp://<http://10.10.10.103:8554/mpeg4ESVideoTest>10.10.10.103:8554/mpeg4E
SVideoTest"
>
>"mpeg1or2AudioVideoTest" stream, from the file "test.mpg"
>Play this stream using the URL
>"rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/
mpeg1or2AudioVideoTest"
>
>Is it the URL link to use with openRTSP client, that I ask below?
Yes, but you need to have the appropriate file - e.g., a MPEG-1 or 2
Program Stream file named "test.mpg" - in the same directory.
>
>Thirdly, I run the RTSP client, by the command : ./openRTSP, and I
>see the usage instructions :
>
>Usage : ./openRTSP [-p <startPortNum>] [-r|-q|-4|-i] [-a|-v] [-V]
>[-d <duration>] [-D <max-inter-packet-gap-time> [-c] [-S <offset>]
>[-n] [-O] [-t|-T <http-port>] [-u <username> <password>] [-s
><initial-seek-time>] [-z <scale>] [-w <width> -h <height>] [-f
><frames-per-second>] [-y] [-H] [-Q [<measurement-interval>]] [-F
><filename-prefix>] [-b <file-sink-buffer-size>] [-B
><input-socket-buffer-size>] [-I <input-interface-ip-address>] [-m]
><url> (or ./openRTSP -o [-V] <url>)
>
> I have tried to use the bold part, for instance ./openRTSP -o [-V]
>rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/m
peg1or2AudioVideoTest
No, just run
./openRTSP
rtsp://<http://10.10.10.103:8554/mpeg1or2AudioVideoTest>10.10.10.103:8554/mp
eg1or2AudioVideoTest
See the "openRTSP" instructions <http://www.live555.com/openRTSP/> to
understand the command-line options.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080711/60079f15
/attachment-0001.html>
------------------------------
Message: 5
Date: Fri, 11 Jul 2008 07:14:30 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] H264 SDP
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240801c49d1afdcf61@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii"; Format="flowed"
>is your format type (97) correct?
>We use 96
Either is correct (provided that it doesn't clash with another
substream (e.g., audio) that you're also sending.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080711/6369416e
/attachment-0001.html>
------------------------------
Message: 6
Date: Sat, 12 Jul 2008 10:52:11 +1200
From: Ryan Walklin <ryanwalklin at gmail.com>
Subject: [Live-devel] Creating a RTSP stream from DVB-T hardware card
To: live-devel at ns.live555.com
Message-ID: <E00BB502-4510-4A91-B22C-7B9B57E01A30 at gmail.com>
Content-Type: text/plain; charset="us-ascii"; Format="flowed";
DelSp="yes"
Hi,
I'm using an Elgato DVB-T USB tuner (EyeTV for DTT) in combination
with the livemedia library to stream a MPEG2-TS stream containing h264
video and AAC audio (LATM encapsulation) via RTSP over my LAN. I've
based my streaming server on the testMPEG2TransportStreamer sample.
The EyeTV plugin SDK provides a callback which is activated when ~100
or so packets have arrived, and sends the raw TS data via a pipe to
the server code, running in another thread. I've modified the code to
read from the other end of the pipe I created. I've also removed the
MPEG2Framer from the chain as I presumed the packet stream was already
in this format.
This done, I'm able to launch VLC and see the stream for 1-2 seconds,
however VLC proceeds to crash with the console errors:
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (48). 1388 bytes of trailing data
will be dropped!
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (112). 1324 bytes of trailing data
will be dropped!
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (60). 1376 bytes of trailing data
will be dropped!
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size
exceeds the client's buffer size (8). 1428 bytes of trailing data
will be dropped!
From my reading of the mailing list archives, I understand that
somewhere along the encoding chain (Presumably in ByteStreamFileSource
(MultiFramedRTPSource) a buffer is being over-run, but I cannot fathom
where. All the buffers seem to default to ~50-60kb, however from those
errors it seems to be under 100 bytes most of the time. I note the sum
of the streamed+dropped packets is 1440, so presumably this has
something to do with network framing.
I've been playing with this all week and ended up more and more
confused reading the source. I'd apprecitate it if anyone is able to
shed light. I've attached my server thread source for your perusal.
(createRTSPserver is essentially main() from
testtMPEG2TransportStreamer.cpp with my tweaks).
Regards,
Ryan Walklin
-------------- next part --------------
A non-text attachment was scrubbed...
Name: MPEG2RTSPServer.cpp
Type: application/octet-stream
Size: 6079 bytes
Desc: not available
URL:
<http://lists.live555.com/pipermail/live-devel/attachments/20080712/0fac44d3
/attachment.obj>
-------------- next part --------------
- stream structure
- using pipe to copy packets
- raw off card
- how to relay mpeg2 ts as live to rtsp
- buffer size
------------------------------
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
End of live-devel Digest, Vol 57, Issue 13
******************************************
More information about the live-devel
mailing list