[Live-devel] live-devel Digest, Vol 43, Issue 21
singh, Ravinder
ravinder.singh at ti.com
Thu May 31 06:30:43 PDT 2007
Hi Ross
Our company policy doesn't allow external excess to our network sorry
for it :-), but would like to know what can be the problem behind this
incomplete file streaming, as bit band player is streaming File till end
while our openRTSP is not streaming till end.
With our openRTSP test application, Server is exiting after streaming
Half of the stream.
Regards
Ravinder
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Sent: Tuesday, May 29, 2007 4:05 PM
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Subject: live-devel Digest, Vol 43, Issue 21
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When replying, please edit your Subject line so it is more specific
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Today's Topics:
1. Re: Read multiple frames? (Ross Finlayson)
2. Re: Problem with openRTSP and testMPEG4VideoStreamer
(Igor Trevisan)
3. SeqNo (homocean at ibr.cs.tu-bs.de)
4. H264 Framerate (Abe Friesen)
5. Re: SeqNo (Ross Finlayson)
6. How to Stream different media format(mpeg4 & pcm) at the same
time? (???)
7. Re: How to Stream different media format(mpeg4 & pcm) at the
same time? (Ross Finlayson)
8. Re: How can I change the received packet at the client by
using openRTSP? (jerry zhao)
9. stream not played till end (singh, Ravinder)
10. Re: stream not played till end (Ross Finlayson)
----------------------------------------------------------------------
Message: 1
Date: Mon, 28 May 2007 01:53:10 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] Read multiple frames?
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240801c28048e56cca@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>>Therefore, if you are feeding input from a "MPEG1or2VideoRTPSource"
>>into a decoder, and your decoder is not smart enough to decode one
>>slice at a time, then you must aggregate the input data into
>>complete video frames before feeding them to your decoder.
>
>Can this be done with live555 stuff?
No.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
------------------------------
Message: 2
Date: Mon, 28 May 2007 11:03:52 +0200
From: "Igor Trevisan" <igt1972 at gmail.com>
Subject: Re: [Live-devel] Problem with openRTSP and
testMPEG4VideoStreamer
To: "LIVE555 Streaming Media - development & use"
<live-devel at ns.live555.com>
Message-ID:
<5a74a1ca0705280203s5f74bd02x6f53de8b24389de8 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
> You didn't properly read the "openRTSP" documentation (in particular,
> the meaning of the "-o" option)
Thanks Ross! I apologize for my questions: it was a typical case
in which a "RTFM" as answer would have been appropriate! ;)
I.
--
"Much less doesn't mean zero"
-- E.Benetti --
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Message: 3
Date: Mon, 28 May 2007 23:43:26 +0200 (CEST)
From: homocean at ibr.cs.tu-bs.de
Subject: [Live-devel] SeqNo
To: live-devel at ns.live555.com
Message-ID:
<1878.134.169.173.74.1180388606.squirrel at mail.ibr.cs.tu-bs.de>
Content-Type: text/plain;charset=iso-8859-1
Hello *,
I have a question regarding the interface between the live
library and mplayer. As expected, in the case of bigger frames, they are
split in more RTP packages. These packages, are put together in live and
assembled in a bigger package, that contains the whole frame. How could
I
have access from mplayer (ReadBufferQueue class, of the demux_rtp.cpp)
to
the number of rtp packages each frame has been composed from?
I can tell if there is loss at mplayer level by inspecting the last
sequence number from bufferQueue->rtpSource->curPacketRTPSeqNum() before
and after calling the getBuffer(...) in the demux_rtp_fill_buffer
function
of demux_rtp.cpp file. But I can not know if there are multiple frames
lost or just one. If I could know how many RTPpackets each frame is
composed on at live level it would be enough to find out how many frames
were lost.
Any information would be helpful.
Regards,
Silviu Homoceanu.
------------------------------
Message: 4
Date: Mon, 28 May 2007 16:52:10 -0700
From: "Abe Friesen" <abefriesen at itiva.com>
Subject: [Live-devel] H264 Framerate
To: "'LIVE555 Streaming Media - development & use'"
<live-devel at ns.live555.com>
Message-ID: <008301c7a183$35bc0250$4807a8c0 at XENOCIDE>
Content-Type: text/plain; charset="us-ascii"
Is it possible to determine the frame rate of an H264 encoded asset
(encoded
as an Annex B bitstream (NAL units preceded by 0x00 0x00 0x00 0x01))
from
the data stream so that I can accurately set fDurationInMicroseconds in
the
framer?
Also, from the H264 spec it looks like the only way to determine whether
the
current NAL unit ends an access unit is by examining the following NAL
unit
to determine if it starts an access unit - or am I missing something?
Thanks,
Abe Friesen
------------------------------
Message: 5
Date: Mon, 28 May 2007 17:33:16 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] SeqNo
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240802c28122a3400f@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>But I can not know if there are multiple frames
>lost or just one. If I could know how many RTPpackets each frame is
>composed on at live level it would be enough to find out how many
frames
>were lost.
As you noted, the "RTPSource" abstraction delivers complete 'frames'.
(However, the term 'frames' is somewhat of a misnomer, because they
do not always correspond to entire video frames - e.g., for MPEG-1 or
2, it delivers complete 'slices'; for H.264, it delivers complete NAL
units). The number of RTP packets that made up each frame is
(deliberately) not exposed outside the "RTPSource" interface. Data
receivers (e.g., audio or video decoders) should not care about this
information.
However, if you just want to find out packet loss rates, then you can
do so by looking at the "RTPReceptionStats" data. (Alternatively,
the data sender (i.e., server) can look at RTCP Reception Report
("RR") coming back from the receiver; this data is recorded in a
"RTPTransmissionStats" object.)
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
------------------------------
Message: 6
Date: Tue, 29 May 2007 12:03:36 +0900
From: ??? <jclee at micro-web.co.kr>
Subject: [Live-devel] How to Stream different media format(mpeg4 &
pcm) at the same time?
To: "live Media" <live-devel at ns.live555.com>
Message-ID: <FMELKCHNDJKAPNMJJJLFEECJCAAA.jclee at micro-web.co.kr>
Content-Type: text/plain; charset="ks_c_5601-1987"
Hi~ ^^
Thanks for your answer~ ^^
testonDemandRTSPServer can stream multiple media files at the same
time,then how can I use/modify the testOnDemandRTSPServer or livemedia
server to stream Mpeg4 video & pcm (wave) audio at the same time to
single client(just one connection), i.e how can I multiplex & stream
separate mpeg4 video data & PCM (wave) audio data.?.. is it possible?
Actually our system have seperate stream for mpeg4 video & PCM audio
which I wanna stream using livemedia to play at the same time by
single media player.
Using livemedia source Im able to stream mpeg4 video & pcm wave audio
seperately . but how can i stream them together to play at the same
time by same player.
------------------------------
Message: 7
Date: Mon, 28 May 2007 23:06:59 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] How to Stream different media format(mpeg4 &
pcm) at the same time?
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240803c28173346888@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii" ; format="flowed"
>testonDemandRTSPServer can stream multiple media files at the same
>time,then how can I use/modify the testOnDemandRTSPServer or
>livemedia server to stream Mpeg4 video & pcm (wave) audio at the
>same time to single client(just one connection), i.e how can I
>multiplex & stream separate mpeg4 video data & PCM (wave) audio
>data.?.. is it possible?
Yes. You would create a "ServerMediaSession" object, and add two
separate "ServerMediaSubsession" objects to it - one for the MPEG-4
video stream, one for the PCM audio stream.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
------------------------------
Message: 8
Date: Tue, 29 May 2007 10:55:40 +0200
From: "jerry zhao" <zhj.zhao at gmail.com>
Subject: Re: [Live-devel] How can I change the received packet at the
client by using openRTSP?
To: live-devel at ns.live555.com
Message-ID:
<b9f77bff0705290155k4f2a4ffcyfaf4f7da6eff9644 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
>>I used the testOnDemandRTSPServer to stream video und openRTSP to
>>receive the streamed data. My question is: where I should change the
>>received packet data before the client stores the received packet on
>>the disk, e.g. adding some data or a special header to the received
>>packets.
>You would need to insert a new filter object (i.e., a subclass of
>"FramedFilter" - that you would write), and insert it in the data
>chain, in front of the "FileSink" object. (You would do this in
>"testProgs/playCommon.cpp".)
Hi, Ross,
thanks for your reply. If one frame consists of several packets and I
need
to
add some additional data to this frame until all the packets of this
frame
are
received, can I still implement this by writing a subclass of
FramedFilter?
Is there any reference code for this kind of implementation?
Thanks again.
Jerry
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Message: 9
Date: Tue, 29 May 2007 14:50:25 +0530
From: "singh, Ravinder" <ravinder.singh at ti.com>
Subject: [Live-devel] stream not played till end
To: <live-devel at ns.live555.com>
Message-ID: <7EAD1AEEA7621C45899FE99123E124A0DA1AFF at dbde01.ent.ti.com>
Content-Type: text/plain; charset="us-ascii"
Hi All
I am trying to use your openRTSP application to stream data from our Bit
Band VOD server, what I have
Observed is openRTSP never streams complete video data, if the file is
around 50MB only 16MB is streamed,
Would like to know why this is happening.
I am writing to file, and using following command
. /openRTSP -V -Q
"rtsp://172.24.141.104:554/Video/nature_mp_544x480_2000.ts"
Server is correctly reporting npt time (Total Duration of Call) as
a=range: npt=0.0-356.774, while only 60 sec streams
Is streamed.
regards
Ravinder
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Message: 10
Date: Tue, 29 May 2007 03:34:52 -0700
From: Ross Finlayson <finlayson at live555.com>
Subject: Re: [Live-devel] stream not played till end
To: LIVE555 Streaming Media - development & use
<live-devel at ns.live555.com>
Message-ID: <f06240807c281b225b408@[66.80.62.44]>
Content-Type: text/plain; charset="us-ascii"
>I am trying to use your openRTSP application to stream data from our
>Bit Band VOD server, what I have
>Observed is openRTSP never streams complete video data, if the file
>is around 50MB only 16MB is streamed,
>Would like to know why this is happening.
>I am writing to file, and using following command
>. /openRTSP -V -Q
"rtsp://172.24.141.104:554/Video/nature_mp_544x480_2000.ts"
Sorry, but your host "172.24.141.104" is not reachable from our
network, therefore I can't test this for myself.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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