From jiri.pinkava at vscht.cz Sat Jul 1 13:45:12 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Sat, 1 Jul 2006 22:45:12 +0200
Subject: [Live-devel] MPEG2TrasportStreamFramer 100% CPU usage
(MPEG2TrasnsportStreamFramerSimple)
Message-ID: <20060701224512.3be69144.jiri.pinkava@vscht.cz>
Hi,
I will make it quick. For some reasons I create new TS stream framer. Goal is be more robust. It does not include any packet timing routines (like computation fDurationInMicroseconds and presentationTime). If there will be interest from live555 side I may create some derived class which do this job (similar/same thing like is now in MPEG2TrasportStreamFramer). All comments/questions/suggestions are welcomed.
Currently present MPEG2TrasportStreamFramer work (in many cases) well and this is new code which may bring new bugs, i will not be disappointed if this code will be refused.
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From zhangzx at rcs-9000.com Sun Jul 2 17:13:51 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Mon, 3 Jul 2006 08:13:51 +0800
Subject: [Live-devel] problems with new version
Message-ID: <000001c69e35$93204a20$347457c6@rcs9000.com>
Dear Ross, I have submitted my problem days ago. But last time I didnt
explain it clearly.
When my application is built with live.2006.01.05, it works fairly well
when I issued "PAUSE" and "RESUME" command. But when I update live555 to
live.2006.05.17 to support RTP over both TCP and UDP, problems happened.
When streaming with unicast I issued "PAUSE", the program is frozen at
singlestep(). But this functions has functioned very well with
live.2006.01.05 version.
Also streaming with TCP "PAUSE" and "RESUME" cann't functions correctly.
I have tried my best to test it, but failed.:(
And I need your help very much, I submitted my probem again.
Can you tell me possible reason and possible solution?
Thanks a lot.
Zhixue Zhang
Nari-relays Electric Corporation, Ltd.
Tel:025-52100626
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From finlayson at live555.com Sun Jul 2 18:57:07 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 02 Jul 2006 18:57:07 -0700
Subject: [Live-devel] MPEG2TrasportStreamFramer 100% CPU usage
(MPEG2TrasnsportStreamFramerSimple)
In-Reply-To: <20060701224512.3be69144.jiri.pinkava@vscht.cz>
References: <20060701224512.3be69144.jiri.pinkava@vscht.cz>
Message-ID: <7.0.1.0.1.20060702185259.01fefb08@live555.com>
>I will make it quick. For some reasons I create new TS stream
>framer. Goal is be more robust. It does not include any packet
>timing routines (like computation fDurationInMicroseconds and
>presentationTime).
These things are important; they should not be omitted.
(In particular, if you are streaming from a file, then
"fDurationInMicroseconds" tells the downstream object how long to
wait between requesting each new Transport Stream packet. If you
omit this (or set it to zero), then the file will be streamed out at
full speed, regardless of the stream's actual bitrate.)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From zhangzx at rcs-9000.com Sun Jul 2 19:24:11 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Mon, 3 Jul 2006 10:24:11 +0800
Subject: [Live-devel] problems with new version
In-Reply-To: <000001c69e35$93204a20$347457c6@rcs9000.com>
Message-ID: <000001c69e47$c56d7770$347457c6@rcs9000.com>
This is no answer for my problem. Didn't I explain it correctly or
could't it happen?
-----????-----
???: live-devel-bounces at ns.live555.com [mailto:live-devel-bounces at ns.
live555.com] ?? Zhixue Zhang
????: 2006?7?3? 8:14
???: live-devel at ns.live555.com
??: [Live-devel] problems with new version
Dear Ross, I have submitted my problem days ago. But last time I didnt
explain it clearly.
When my application is built with live.2006.01.05, it works fairly well
when I issued "PAUSE" and "RESUME" command. But when I update live555 to
live.2006.05.17 to support RTP over both TCP and UDP, problems happened.
When streaming with unicast I issued "PAUSE", the program is frozen at
singlestep(). But this functions has functioned very well with
live.2006.01.05 version.
Also streaming with TCP "PAUSE" and "RESUME" cann't functions correctly.
I have tried my best to test it, but failed.:(
And I need your help very much, I submitted my probem again.
Can you tell me possible reason and possible solution?
Thanks a lot.
Zhixue Zhang
Nari-relays Electric Corporation, Ltd.
Tel:025-52100626
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From finlayson at live555.com Sun Jul 2 21:04:21 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 02 Jul 2006 21:04:21 -0700
Subject: [Live-devel] problems with new version
In-Reply-To: <000001c69e47$c56d7770$347457c6@rcs9000.com>
References: <000001c69e35$93204a20$347457c6@rcs9000.com>
<000001c69e47$c56d7770$347457c6@rcs9000.com>
Message-ID: <7.0.1.0.1.20060702205945.02006d68@live555.com>
>This is no answer for my problem.
This is a public mailing list that anyone (including non-customers)
can post to. However, not every posted question will get an answer.
In your case - I don't have time to look at your problem right
now. A change made a few months ago should have prevented any
problems with "PAUSE" and RTP-over-TCP, so I don't know why you might
still be experiencing problems with this. For now, you're going to
have to look at this yourself.
Remember, You Have Complete Source Code.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From clem.taylor at gmail.com Sun Jul 2 23:42:24 2006
From: clem.taylor at gmail.com (Clem Taylor)
Date: Mon, 3 Jul 2006 02:42:24 -0400
Subject: [Live-devel] use with uclibc
In-Reply-To:
References:
Message-ID:
On 6/18/06, Lees, Christian wrote:
> Has anyone gotten the LIVE libraries to work under a system using uClibc. I
> have managed to get the libraries and test programs to compile, but when I
> go to run the test programs I get the warnings
I've been using liveMedia with uClibc for almost a year without much
trouble. Are you sure you have multicast in your kernel? I think you
might get that error if multicast isn't compiled in. Check for
CONFIG_IP_MULTICAST in your kernel .config.
--Clem
From max at code-it-now.com Mon Jul 3 02:11:28 2006
From: max at code-it-now.com (Maxim Petrov)
Date: Mon, 03 Jul 2006 12:11:28 +0300
Subject: [Live-devel] RTP timestamp
Message-ID: <44A8DF40.4040207@code-it-now.com>
Hi Ross!
Is it possible to attach any timestamp to frame before sending to client?
For example we have some JPEG files. These files was stored in Juny 1
2006 12:30:43.10 from Axis webcamera.
These files stored as archive with random access.
if I'm streaming video from that archive (as sequences of JPEG files)
then frame time must be corresponded to archive time of the frame.
I tried to play with fPresentationTime variable. But seems is not what I
need, cause on client side (Filesink::addData) presentationTime value is
always current time.
--
Bye, Maxim.
From yoann.ramard at orange-ft.com Mon Jul 3 03:01:39 2006
From: yoann.ramard at orange-ft.com (zze-RAMARD Yoann RD-TECH-GRE)
Date: Mon, 3 Jul 2006 12:01:39 +0200
Subject: [Live-devel] Switching sources
Message-ID:
> Hi,
> I'd like to change the source without disconnecting the destination.
> So I did sink->stopPlaying() then sink->startPlaying(newSource). The
> sources are mp3 streams (not files). But when this change happens the
> stream I got is not changing the way I expect. There is a one second
> normal play then a jump forward in the song and then the music is bad
> quality and jerky. Everything goes back normal if stop then start
> again VLC (but it's not the way I want it to work). Does anybody knows
> where the problem come from? My opinion is a buffer problem. Thanks
> for your answers.
>
> Yoann RAMARD
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From finlayson at live555.com Mon Jul 3 10:37:34 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 03 Jul 2006 10:37:34 -0700
Subject: [Live-devel] RTP timestamp
In-Reply-To: <44A8DF40.4040207@code-it-now.com>
References: <44A8DF40.4040207@code-it-now.com>
Message-ID: <7.0.1.0.1.20060703103423.01feae68@live555.com>
>Is it possible to attach any timestamp to frame before sending to client?
>For example we have some JPEG files. These files was stored in Juny 1
>2006 12:30:43.10 from Axis webcamera.
>These files stored as archive with random access.
>
>if I'm streaming video from that archive (as sequences of JPEG files)
>then frame time must be corresponded to archive time of the frame.
>
>I tried to play with fPresentationTime variable. But seems is not what I
>need, cause on client side (Filesink::addData) presentationTime value is
>always current time.
The "fPresentationTime" variable that is set by a source object must
be aligned with 'wall clock time' - i.e. the time that you would get
by calling "gettimeofday()". Therefore, if your original timestamps
come from some other time base, you must first add or subtract an
appropriate offset before you assign it to "fPresentationTime".
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Mon Jul 3 10:42:29 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 03 Jul 2006 10:42:29 -0700
Subject: [Live-devel] Switching sources
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060703103745.0202f020@live555.com>
>I'd like to change the source without disconnecting the destination.
>So I did sink->stopPlaying() then sink->startPlaying(newSource). The
>sources are mp3 streams (not files). But when this change happens
>the stream I got is not changing the way I expect.
I think a better way to do this is to retain the same source object,
but to change the underlying media source that it reads from. For
example, if you are currently reading from a "ByteStreamFileSource",
then I suggest using a new source class (that you would write) that
lets you change the underlying input 'open file'. That way, you
don't need to mess around with "stopPlaying()"/"startPlaying()" - the
RTP/RTCP-related code need not know that input source has changed.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From zhangzx at rcs-9000.com Mon Jul 3 20:21:23 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Tue, 4 Jul 2006 11:21:23 +0800
Subject: [Live-devel] problems with new version
In-Reply-To: <7.0.1.0.1.20060702205945.02006d68@live555.com>
Message-ID: <000001c69f18$f0534ba0$347457c6@rcs9000.com>
My problem is partly sovled by commenting "/*fTCPStreamIdCount == */" in
RTSPClient::playMediaSession().
In fact, I dont understand it clearly.
New probelm arised. When streaming with TCP, server dumped after a
while.
Can anyone give any advices?
This is a public mailing list that anyone (including non-customers)
can post to. However, not every posted question will get an answer.
In your case - I don't have time to look at your problem right
now. A change made a few months ago should have prevented any
problems with "PAUSE" and RTP-over-TCP, so I don't know why you might
still be experiencing problems with this. For now, you're going to
have to look at this yourself.
Remember, You Have Complete Source Code.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
From sdhays.neon.com.tw at gmail.com Mon Jul 3 23:06:53 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Tue, 4 Jul 2006 14:06:53 +0800
Subject: [Live-devel] OnDemandServerMediaSubsession bug
Message-ID: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.com>
I've encountered the following situation when using reuseFirstSource=True
doing on-demand streaming: if the stream source ends and calls
handleClosure(this), the streamState reference count is decremented to 0
BEFORE OnDemandServerMediaSubsession::deleteStream() is called. So
deleteStream() doesn't delete the streamState because the reference count is
not > 0 and nothing else deletes it either. This causes the first time a
client reconnects after a source ends to fail because the server is trying
to use a non-existent source. The second attempt works fine because the
failed try increments the reference count and then allows deleteStream to
delete the streamState properly.
The fix I've written for this is to change lines 303 and 304 in
OnDemandServerMediaSubsession.cpp to:
if (streamState != NULL && streamState ->referenceCount() >= 0) {
if (0 < streamState->referenceCount()) --streamState->referenceCount();
I'm not sure if this is the best way to fix this, but it's working for me.
This situation occurs for me when using a source class that I have
implemented whose media source (an encoder) sometimes needs to restart,
causing the source class to signal end-of-stream using handleClosure(this).
It also occurs in testOnDemandRTSPServer when reuseFirstSource is set to
"True".
Scott
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From sdhays.neon.com.tw at gmail.com Mon Jul 3 23:20:52 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Tue, 4 Jul 2006 14:20:52 +0800
Subject: [Live-devel] OnDemandServerMediaSubsession bug
In-Reply-To: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.com>
References: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.com>
Message-ID: <9866ce4f0607032320x6a64354amebb64c2bd47a32dc@mail.gmail.com>
I forgot to mention that this problem occurs in version 2006.07.04 (although
I first discovered it in 2006.05.17).
Scott
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From zhangzx at rcs-9000.com Mon Jul 3 23:50:04 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Tue, 4 Jul 2006 14:50:04 +0800
Subject: [Live-devel] rtp over tcp
In-Reply-To: <000001c69f18$f0534ba0$347457c6@rcs9000.com>
Message-ID: <000001c69f36$177a01c0$347457c6@rcs9000.com>
Hi, everyone
LIVE555 Streaming Media is built on SOLARIS 9.
"testOnDemandRTSPServer" and "openRTSP" are employed to test treaming
with "RTP over TCP".
When streaming with "RTP over TCP", after a while openRTSP exited, and
"Broken pipe" was reported.
I have test many times with the same test.m4v, and file openRTSP saved
is the same size. It seemed that some media bytes cann't be delivered.
Other m4v file is tested also, the result is the same.
Did anyone have some problems?
Zhixue Zhang
From sdhays.neon.com.tw at gmail.com Tue Jul 4 00:06:26 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Tue, 4 Jul 2006 15:06:26 +0800
Subject: [Live-devel] rtp over tcp
In-Reply-To: <000001c69f36$177a01c0$347457c6@rcs9000.com>
References: <000001c69f18$f0534ba0$347457c6@rcs9000.com>
<000001c69f36$177a01c0$347457c6@rcs9000.com>
Message-ID: <9866ce4f0607040006j6622de92sb41ff84be17f648e@mail.gmail.com>
Is your client on the same host?
I just saw this comment by some defined-out code in RTSPServer.cpp when I
was looking for the source of a bug:
// Ignore the SIGPIPE signal, so that clients on the same host that are
killed don't also kill us:
That sounds like what you're experiencing if openRTSP is also on the same
host. It requires enabling the preprocessor define: USE_SIGNALS and then
recompiling the library.
Hope that helps,
Scott
On 7/4/06, Zhixue Zhang wrote:
>
> Hi, everyone
> LIVE555 Streaming Media is built on SOLARIS 9.
> "testOnDemandRTSPServer" and "openRTSP" are employed to test treaming
> with "RTP over TCP".
> When streaming with "RTP over TCP", after a while openRTSP exited, and
> "Broken pipe" was reported.
>
>
> I have test many times with the same test.m4v, and file openRTSP saved
> is the same size. It seemed that some media bytes cann't be delivered.
> Other m4v file is tested also, the result is the same.
>
> Did anyone have some problems?
>
>
> Zhixue Zhang
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From zhangzx at rcs-9000.com Tue Jul 4 00:50:45 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Tue, 4 Jul 2006 15:50:45 +0800
Subject: [Live-devel] rtp over tcp
In-Reply-To: <9866ce4f0607040006j6622de92sb41ff84be17f648e@mail.gmail.com>
Message-ID: <001101c69f3e$91952fe0$347457c6@rcs9000.com>
Thanks a lot for your help.
According to your advice, I test it on different hosts.
testOnDemandRTSP run on Windoze, and openRTSP run on SOLARIS.
I got the same result. Streaming is broken after a while also.
Zhixue Zhang
I s your client on the same host?
I just saw this comment by some defined-out code in RTSPServer.cpp when
I was looking for the source of a bug:
// Ignore the SIGPIPE signal, so that clients on the same host that are
killed don't also kill us:
That sounds like what you're experiencing if openRTSP is also on the
same host. It requires enabling the preprocessor define: USE_SIGNALS
and then recompiling the library.
Hope that helps,
Scott
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From zhangzx at rcs-9000.com Tue Jul 4 01:57:12 2006
From: zhangzx at rcs-9000.com (Zhixue Zhang)
Date: Tue, 4 Jul 2006 16:57:12 +0800
Subject: [Live-devel] rtp over tcp
In-Reply-To: <001101c69f3e$91952fe0$347457c6@rcs9000.com>
Message-ID: <000001c69f47$da62a4b0$347457c6@rcs9000.com>
I have found the reason with "RTP over TCP". In SingleStep(),
WSAGetLastError() returned 10038.
So exit(0) is called.
But I dont know how to deal with it.
Any advices?
Zhixue Zhang
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From finlayson at live555.com Tue Jul 4 02:46:18 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 04 Jul 2006 02:46:18 -0700
Subject: [Live-devel] OnDemandServerMediaSubsession bug
In-Reply-To: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.co
m>
References: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060704023304.01fe8240@live555.com>
>I've encountered the following situation when using
>reuseFirstSource=True doing on-demand streaming: if the stream
>source ends and calls handleClosure(this), the streamState reference
>count is decremented to 0 BEFORE
>OnDemandServerMediaSubsession::deleteStream() is called.
I don't see how this can happen, because
"OnDemandServerMediaSubsession::deleteStream() " is the only place
where the reference count is *decremented*. I assume, therefore,
that you are instead referring to the reference count being *set*
(not decremented) to 0 in "StreamState::reclaim()" (which is called
by "afterPlayingStreamState()")?
> So deleteStream() doesn't delete the streamState because the
> reference count is not > 0 and nothing else deletes it
> either. This causes the first time a client reconnects after a
> source ends to fail because the server is trying to use a
> non-existent source. The second attempt works fine because the
> failed try increments the reference count and then allows
> deleteStream to delete the streamState properly.
>
>The fix I've written for this is to change lines 303 and 304 in
>OnDemandServerMediaSubsession.cpp to:
> if (streamState != NULL && streamState ->referenceCount() >= 0) {
> if (0 < streamState->referenceCount()) --streamState->referenceCount();
I *think* this will work, although I'm a bit worried about the
"StreamState" object getting deleted twice (once when the reference
count was 1, and again when the reference count was 0). However,
because the "streamToken" reference parameter will get set to NULL
the first time the object gets deleted, I don't think that situation
will occur.
So, I'll go ahead and make this change in the next release of the
software. In the meantime, please let me know if it causes any
problems for you.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From sdhays.neon.com.tw at gmail.com Tue Jul 4 04:04:22 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Tue, 4 Jul 2006 19:04:22 +0800
Subject: [Live-devel] OnDemandServerMediaSubsession bug
In-Reply-To: <7.0.1.0.1.20060704023304.01fe8240@live555.com>
References: <9866ce4f0607032306p1373637duc04fbb591e2ccb31@mail.gmail.com>
<7.0.1.0.1.20060704023304.01fe8240@live555.com>
Message-ID: <9866ce4f0607040404u3c93d9e1nd2a4e137721a53b4@mail.gmail.com>
On 7/4/06, Ross Finlayson wrote:
>
>
> I don't see how this can happen, because
> "OnDemandServerMediaSubsession::deleteStream() " is the only place
> where the reference count is *decremented*. I assume, therefore,
> that you are instead referring to the reference count being *set*
> (not decremented) to 0 in "StreamState::reclaim()" (which is called
> by "afterPlayingStreamState()")?
That's right. deleteStream is getting called while
streamState->referenceCount() is already 0, so it doesn't enter the if
statement that deletes streamState and sets streamToken to NULL. This
causes fLastStreamToken to not get set to NULL, so the next time
getStreamParameters() gets called, it takes the "else" path instead of
taking the "first source" path.
I *think* this will work, although I'm a bit worried about the
> "StreamState" object getting deleted twice (once when the reference
> count was 1, and again when the reference count was 0). However,
> because the "streamToken" reference parameter will get set to NULL
> the first time the object gets deleted, I don't think that situation
> will occur.
The count seems to be set to 0 in StreamState::afterPlayingStreamState()
(which calls streamState->reclaim()) first, then deleteStream() is called.
So it seems to still only be getting deleted once: in deleteStream().
So, I'll go ahead and make this change in the next release of the
> software. In the meantime, please let me know if it causes any
> problems for you.
Thanks,
Scott
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From max at code-it-now.com Tue Jul 4 00:55:04 2006
From: max at code-it-now.com (Maxim)
Date: Tue, 4 Jul 2006 10:55:04 +0300
Subject: [Live-devel] RTP timestamp
References: <44A8DF40.4040207@code-it-now.com>
<7.0.1.0.1.20060703103423.01feae68@live555.com>
Message-ID: <00ba01c69f3f$28ef1130$8000a8c0@maxvm>
Hi Ross!
> The "fPresentationTime" variable that is set by a source object must
> be aligned with 'wall clock time' - i.e. the time that you would get
> by calling "gettimeofday()". Therefore, if your original timestamps
> come from some other time base, you must first add or subtract an
> appropriate offset before you assign it to "fPresentationTime".
>
It's what excactly I tried to do. But seems I'm not enough understand this
mechanism cause it will not work for me.
My code on server side looks like:
void MyFileSource::incomingPacketHandler1() {
// Getting jpeg frame from archive
// Getting its timestamp
// Assigning this timestamp to "fPresentationTime"
fPresentationTime = currentFrameTimestamp;
// Delivering frame to client, calling "afterGetting"
}
Is there any wrong?
But on client side (Filesink::addData) "presentationTime" always equals
client's local timestamp. And increments when new frame received.
Great thanks for advance!
From glen at lincor.com Tue Jul 4 08:25:24 2006
From: glen at lincor.com (Glen Gray)
Date: Tue, 04 Jul 2006 16:25:24 +0100
Subject: [Live-devel] [PATCH]MPEG1-Audio for Kasenna
Message-ID: <44AA8864.70509@lincor.com>
Hey Ross/Guys,
Here is a set of 3 patches we've been working on, to enable MPEG-Audio
playback from Kasenna.
live_mpa_sdp_fix.diff:: modified RTSPClient.cpp to allow it to generate
a valid SDP string from Kasenna's Description data. Previously it was
hardcoded to just generate a Video SDP. This was done by me back in March.
The next 2 patches where made by Greg Farrell . If
these are accepted, please ensure he gets the credit for them.
live_mpeg_audio.diff:: this is effectively the final part of the above
patch and was implemented by Greg after I moved off the project onto
something else. This basically wraps the in coming UDP data with
MPEG1or2AudioStreamFramer, based on the fCodeName string, in much the
same way that MPEG2TransportStreamFramer was called to enable Mpeg2-ts
video playback from the Kasenna.
With the first patch above, we got our version of VLC establishing and
playing back an MPEG1-Audio stream from our Kasenna, however we didn't
get to hear the Audio (no pts's on the raw data). With the second patch
we could hear the audio, but noticed a problem. After a while we lost
sync and eventually the stream collapsed.
live555_udp.diff::
Quote from Greg
> The StreamParser class uses two receive buffers or 'banks'. It receives data into
> one bank until it is full then swaps to the other bank and so on.
>
> How the StreamParser class decides if there is enough room in the current buffer
> to perform a receive is by checking if the space left is greater than the number
> of bytes of data requested from the StreamParser.
>
> However if the underlying data source is UDP (BasicUdpSource) then a receive with
> a max length of less than the size of incoming UDP packets will result in the rest of
> the packet being silently dropped in GroupsockHelper/readSocket().
>
> This will happen whenever there is enough room for a frame (the length of data
> StreamParser will be asked for) in a bank buffer but not enough room for a full
> UDP packet. Kasenna's will put multiple mpeg audio frames in one UDP packet
> for efficiency I assume and this will result in lost data on most bank buffer swaps.
>
> To fix it I've changed the StreamParser to swap banks whenever it believes there is
> not enough room left for a packet as well as a frame. As you can't tell in advance what
> size incoming UDP packets will be the StreamParser defaults to a guess of 512 bytes
> and increases that whenever it sees larger packets.
We are still experiencing some problems however. Namely, pts drifts out
of sync and eventually the streams break down. We'd put a hack in place
to force the pts values to stay synced and that was when we noticed that
we where loosing UDP data. Fixing that with the final patch listed
above, we've put all these patches into our VLC build and noticed that
we are still experiencing pts drift (which we'd assumed was happening
because of the lost UDP data).
We're continuing to investigate.
Kind regards,
--
Glen Gray Digital Depot, Thomas Street
Senior Software Engineer Dublin 8, Ireland
Lincor Solutions Ltd. Ph: +353 (0) 1 4893682
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From finlayson at live555.com Tue Jul 4 08:45:40 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 04 Jul 2006 08:45:40 -0700
Subject: [Live-devel] RTP timestamp
In-Reply-To: <00ba01c69f3f$28ef1130$8000a8c0@maxvm>
References: <44A8DF40.4040207@code-it-now.com>
<7.0.1.0.1.20060703103423.01feae68@live555.com>
<00ba01c69f3f$28ef1130$8000a8c0@maxvm>
Message-ID: <7.0.1.0.1.20060704084004.01fe8240@live555.com>
>My code on server side looks like:
>
>void MyFileSource::incomingPacketHandler1() {
>// Getting jpeg frame from archive
>// Getting its timestamp
>// Assigning this timestamp to "fPresentationTime"
>fPresentationTime = currentFrameTimestamp;
>
>// Delivering frame to client, calling "afterGetting"
>}
>Is there any wrong?
Yes - you didn't read my previous email.
You can't just set "fPresentationTime" to
"currentFrameTimestamp". Instead, you must first add or subtract an
offset so that it is aligned with 'wall clock time'.
I.e., for the first timestamp (in pseudo code):
gettimeofday(&timeNow, NULL)
offset = timeNow - currentFrameTimestamp;
fPresentationTime = timeNow;
For the second and subsequent timestamps:
fPresentationTime = currentFrameTimestamp - offset;
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From jiri.pinkava at vscht.cz Tue Jul 4 11:18:58 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Tue, 4 Jul 2006 20:18:58 +0200
Subject: [Live-devel] MPEG2TrasportStreamFramer 100% CPU usage
(MPEG2TrasnsportStreamFramerSimple)
In-Reply-To: <7.0.1.0.1.20060702185259.01fefb08@live555.com>
References: <20060701224512.3be69144.jiri.pinkava@vscht.cz>
<7.0.1.0.1.20060702185259.01fefb08@live555.com>
Message-ID: <20060704201858.37c4f97d.jiri.pinkava@vscht.cz>
On Sun, 02 Jul 2006 18:57:07 -0700
Ross Finlayson wrote:
>
> >I will make it quick. For some reasons I create new TS stream
> >framer. Goal is be more robust. It does not include any packet
> >timing routines (like computation fDurationInMicroseconds and
> >presentationTime).
>
> These things are important; they should not be omitted.
>
> (In particular, if you are streaming from a file, then
> "fDurationInMicroseconds" tells the downstream object how long to
> wait between requesting each new Transport Stream packet. If you
> omit this (or set it to zero), then the file will be streamed out at
> full speed, regardless of the stream's actual bitrate.)
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
Well, I now do backward compatibility MPEG2TransportStreamFramer and repair some pretty stupid bugs in my new TS Framer and send this. For live source is necessary do caching (read as much data as possible). Is there a tip how to do this for device/socket source (it is necessary create buffer in SocketSource class or is there way to use buffer of "client")
From braymond at echostorm.net Tue Jul 4 20:56:16 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Tue, 4 Jul 2006 23:56:16 -0400
Subject: [Live-devel] Darwin Injector
Message-ID:
I tried the MPEG4 darwin injector test program to see how it would work
against my DSS server and I didn't have much luck getting it to work.
When live issues the PLAY command DSS returns an error stating a
precondition failed. I've using the latest live release and DSS 5.5.
Can someone verify this is supposed to work in the recent live releases?
- Brian
From finlayson at live555.com Tue Jul 4 23:15:04 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 04 Jul 2006 23:15:04 -0700
Subject: [Live-devel] Darwin Injector
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060704231300.020012f0@live555.com>
At 08:56 PM 7/4/2006, you wrote:
>I tried the MPEG4 darwin injector test program to see how it would work
>against my DSS server and I didn't have much luck getting it to work.
>When live issues the PLAY command DSS returns an error stating a
>precondition failed. I've using the latest live release and DSS 5.5.
>
>Can someone verify this is supposed to work in the recent live releases?
I don't know of any problems. Note that the "precondition failed"
error is Darwin bogosity. It seems to occur if you try to inject a
new stream - with the same SDP name as before - too soon after the
last time you tried. Wait a minute or so (for Darwin to time out
some state) and try again.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From glen at lincor.com Wed Jul 5 03:39:45 2006
From: glen at lincor.com (Glen Gray)
Date: Wed, 05 Jul 2006 11:39:45 +0100
Subject: [Live-devel] [PATCH]MPEG1-Audio for Kasenna
In-Reply-To: <44AA8864.70509@lincor.com>
References: <44AA8864.70509@lincor.com>
Message-ID: <44AB96F1.3030409@lincor.com>
Further to my mail last night, Greg found the reason for the continued
issues.
StreamParser.cpp is using the wrong index in
StreamParser::ensureValidBytes1() to check for enough remaining space
in a bank buffer.
- if (fCurParserIndex + bufSpaceNeeded > BANK_SIZE ) {
+ if (fTotNumValidBytes + bufSpaceNeeded > BANK_SIZE ) {
This needs to be combined with my previous UDP data-loss fix
patch to prevent data loss.
This change on it's own will decrease the amount of UDP data
lost but won't stop it.
Greg
--- StreamParser.cpp 2006-07-05 11:29:04.000000000 +0100
+++ StreamParser.cpp.new 2006-07-05 11:29:49.000000000 +0100
@@ -71,7 +71,7 @@
// bank. If so, start using a new bank now.
// if (fCurParserIndex + numBytesNeeded > (BANK_SIZE-MAX_PACKET_SIZE)) {
- if (fCurParserIndex + bufSpaceNeeded > BANK_SIZE ) {
+ if (fTotNumValidBytes + bufSpaceNeeded > BANK_SIZE ) {
// Swap banks, but save any still-needed bytes from the old bank:
unsigned numBytesToSave = fTotNumValidBytes - fSavedParserIndex;
unsigned char const* from = &curBank()[fSavedParserIndex];
Glen Gray wrote:
> Hey Ross/Guys,
>
> Here is a set of 3 patches we've been working on, to enable MPEG-Audio
> playback from Kasenna.
>
> live_mpa_sdp_fix.diff:: modified RTSPClient.cpp to allow it to generate
> a valid SDP string from Kasenna's Description data. Previously it was
> hardcoded to just generate a Video SDP. This was done by me back in March.
>
> The next 2 patches where made by Greg Farrell . If
> these are accepted, please ensure he gets the credit for them.
>
> live_mpeg_audio.diff:: this is effectively the final part of the above
> patch and was implemented by Greg after I moved off the project onto
> something else. This basically wraps the in coming UDP data with
> MPEG1or2AudioStreamFramer, based on the fCodeName string, in much the
> same way that MPEG2TransportStreamFramer was called to enable Mpeg2-ts
> video playback from the Kasenna.
>
> With the first patch above, we got our version of VLC establishing and
> playing back an MPEG1-Audio stream from our Kasenna, however we didn't
> get to hear the Audio (no pts's on the raw data). With the second patch
> we could hear the audio, but noticed a problem. After a while we lost
> sync and eventually the stream collapsed.
>
> live555_udp.diff::
>
> Quote from Greg
>> The StreamParser class uses two receive buffers or 'banks'. It
>> receives data into
>> one bank until it is full then swaps to the other bank and so on.
>>
>> How the StreamParser class decides if there is enough room in the
>> current buffer
>> to perform a receive is by checking if the space left is greater than
>> the number
>> of bytes of data requested from the StreamParser.
>>
>> However if the underlying data source is UDP (BasicUdpSource) then a
>> receive with
>> a max length of less than the size of incoming UDP packets will result
>> in the rest of
>> the packet being silently dropped in GroupsockHelper/readSocket().
>>
>> This will happen whenever there is enough room for a frame (the length
>> of data
>> StreamParser will be asked for) in a bank buffer but not enough room
>> for a full
>> UDP packet. Kasenna's will put multiple mpeg audio frames in one UDP
>> packet
>> for efficiency I assume and this will result in lost data on most bank
>> buffer swaps.
>>
>> To fix it I've changed the StreamParser to swap banks whenever it
>> believes there is
>> not enough room left for a packet as well as a frame. As you can't
>> tell in advance what
>> size incoming UDP packets will be the StreamParser defaults to a guess
>> of 512 bytes
>> and increases that whenever it sees larger packets.
>
> We are still experiencing some problems however. Namely, pts drifts out
> of sync and eventually the streams break down. We'd put a hack in place
> to force the pts values to stay synced and that was when we noticed that
> we where loosing UDP data. Fixing that with the final patch listed
> above, we've put all these patches into our VLC build and noticed that
> we are still experiencing pts drift (which we'd assumed was happening
> because of the lost UDP data).
>
> We're continuing to investigate.
> Kind regards,
>
>
> ------------------------------------------------------------------------
>
> diff -uNr vlc-trunk.orig/live/liveMedia/StreamParser.cpp vlc-trunk.work/live/liveMedia/StreamParser.cpp
> --- vlc-trunk.orig/live/liveMedia/StreamParser.cpp 2006-07-04 14:40:17.000000000 +0100
> +++ vlc-trunk.work/live/liveMedia/StreamParser.cpp 2006-07-04 15:33:30.000000000 +0100
> @@ -41,6 +41,7 @@
> fBank[1] = new unsigned char[BANK_SIZE];
> fCurBankNum = 0;
> fCurBank = fBank[fCurBankNum];
> + MAX_PACKET_SIZE = 512;
> }
>
> StreamParser::~StreamParser() {
> @@ -53,15 +54,23 @@
> // We need to read some more bytes from the input source.
> // First, clarify how much data to ask for:
> unsigned maxInputFrameSize = fInputSource->maxFrameSize();
> + unsigned bufSpaceNeeded;
> if (maxInputFrameSize > numBytesNeeded) numBytesNeeded = maxInputFrameSize;
>
> + // Check if our frame size or expected max packet sizes are the limiting factor
> + // on whether we've enough room in our current bank.
> + if (MAX_PACKET_SIZE > numBytesNeeded)
> + bufSpaceNeeded = MAX_PACKET_SIZE;
> + else
> + bufSpaceNeeded = numBytesNeeded;
> +
> // First, check whether these new bytes would overflow the current
> // bank. If so, start using a new bank now.
> - if (fCurParserIndex + numBytesNeeded > BANK_SIZE) {
> + if (fCurParserIndex + bufSpaceNeeded > BANK_SIZE ) {
> // Swap banks, but save any still-needed bytes from the old bank:
> unsigned numBytesToSave = fTotNumValidBytes - fSavedParserIndex;
> unsigned char const* from = &curBank()[fSavedParserIndex];
> -
> +
> fCurBankNum = (fCurBankNum + 1)%2;
> fCurBank = fBank[fCurBankNum];
> memmove(curBank(), from, numBytesToSave);
> @@ -106,6 +115,9 @@
> << numBytesRead << " bytes; expected no more than "
> << BANK_SIZE - buffer->fTotNumValidBytes << "\n";
> }
> + // Check if we're receiving larger packets than expected
> + if ( numBytesRead > buffer->MAX_PACKET_SIZE)
> + buffer->MAX_PACKET_SIZE = numBytesRead;
>
> unsigned char* ptr = &buffer->curBank()[buffer->fTotNumValidBytes];
> buffer->fTotNumValidBytes += numBytesRead;
> Binary files vlc-trunk.orig/live/liveMedia/.StreamParser.cpp.swp and vlc-trunk.work/live/liveMedia/.StreamParser.cpp.swp differ
> diff -uNr vlc-trunk.orig/live/liveMedia/StreamParser.hh vlc-trunk.work/live/liveMedia/StreamParser.hh
> --- vlc-trunk.orig/live/liveMedia/StreamParser.hh 2006-07-04 14:40:17.000000000 +0100
> +++ vlc-trunk.work/live/liveMedia/StreamParser.hh 2006-07-04 15:19:52.000000000 +0100
> @@ -132,6 +132,7 @@
> // The current position of the parser within the current bank:
> unsigned fCurParserIndex; // <= fTotNumValidBytes
> unsigned char fRemainingUnparsedBits; // in previous byte: [0,7]
> + unsigned MAX_PACKET_SIZE;
>
> // The total number of valid bytes stored in the current bank:
> unsigned fTotNumValidBytes; // <= BANK_SIZE
>
>
> ------------------------------------------------------------------------
>
> diff -uNr live/liveMedia/RTSPClient.cpp live.lincor/liveMedia/RTSPClient.cpp
> --- live/liveMedia/RTSPClient.cpp 2006-01-27 09:14:44.000000000 +0000
> +++ live.lincor/liveMedia/RTSPClient.cpp 2006-04-05 15:14:41.000000000 +0100
> @@ -452,17 +452,33 @@
> unsigned char byte2 = (fServerAddress & 0x0000ff00) >> 8;
> unsigned char byte3 = (fServerAddress & 0x00ff0000) >> 16;
> unsigned char byte4 = (fServerAddress & 0xff000000) >> 24;
> -
> - char const* sdpFmt =
> - "v=0\r\n"
> - "o=NoSpacesAllowed 1 1 IN IP4 %u.%u.%u.%u\r\n"
> - "s=%s\r\n"
> - "c=IN IP4 %u.%u.%u.%u\r\n"
> - "t=0 0\r\n"
> - "a=control:*\r\n"
> - "a=range:npt=0-%llu\r\n"
> - "m=video 1554 RAW/RAW/UDP 33\r\n"
> - "a=control:trackID=%d\r\n";
> +
> + char const* sdpFmt = NULL;
> +
> + if (fKasennaContentType != NULL
> + && strcmp(fKasennaContentType, "MPEG1-Audio") == 0) {
> + sdpFmt =
> + "v=0\r\n"
> + "o=NoSpacesAllowed 1 1 IN IP4 %u.%u.%u.%u\r\n"
> + "s=%s\r\n"
> + "c=IN IP4 %u.%u.%u.%u\r\n"
> + "t=0 0\r\n"
> + "a=control:*\r\n"
> + "a=range:npt=0-%llu\r\n"
> + "m=audio 1554 RAW/RAW/UDP 14\r\n"
> + "a=control:trackID=%d\r\n";
> + } else {
> + sdpFmt =
> + "v=0\r\n"
> + "o=NoSpacesAllowed 1 1 IN IP4 %u.%u.%u.%u\r\n"
> + "s=%s\r\n"
> + "c=IN IP4 %u.%u.%u.%u\r\n"
> + "t=0 0\r\n"
> + "a=control:*\r\n"
> + "a=range:npt=0-%llu\r\n"
> + "m=video 1554 RAW/RAW/UDP 33\r\n"
> + "a=control:trackID=%d\r\n";
> + }
> unsigned sdpBufSize = strlen(sdpFmt)
> + 4*3 // IP address
> + strlen(url)
>
>
> ------------------------------------------------------------------------
>
> diff -uNr vlc-trunk.orig/live/liveMedia/MediaSession.cpp vlc-trunk.work/live/liveMedia/MediaSession.cpp
> --- vlc-trunk.orig/live/liveMedia/MediaSession.cpp 2006-06-21 10:34:35.000000000 +0100
> +++ vlc-trunk.work/live/liveMedia/MediaSession.cpp 2006-06-21 16:30:33.000000000 +0100
> @@ -648,7 +648,6 @@
> env().setResultMsg(tmpBuf);
> break;
> }
> -
> // Check "fProtocolName"
> if (strcmp(fProtocolName, "UDP") == 0) {
> // A UDP-packetized stream (*not* a RTP stream)
> @@ -659,6 +658,9 @@
> fReadSource = MPEG2TransportStreamFramer::createNew(env(), fReadSource);
> // this sets "durationInMicroseconds" correctly, based on the PCR values
> }
> + else if (strcmp(fCodecName, "MPA") == 0) { // MPEG Audio Stream
> + fReadSource =MPEG1or2AudioStreamFramer::createNew(env(), fReadSource);
> + }
> } else {
> // Check "fCodecName" against the set of codecs that we support,
> // and create our RTP source accordingly
--
Glen Gray Digital Depot, Thomas Street
Senior Software Engineer Dublin 8, Ireland
Lincor Solutions Ltd. Ph: +353 (0) 1 4893682
From braymond at echostorm.net Wed Jul 5 05:13:01 2006
From: braymond at echostorm.net (Brian Raymond)
Date: Wed, 5 Jul 2006 08:13:01 -0400
Subject: [Live-devel] Darwin Injector
In-Reply-To: <7.0.1.0.1.20060704231300.020012f0@live555.com>
References:
<7.0.1.0.1.20060704231300.020012f0@live555.com>
Message-ID: <95e8ca806633be8b56809233190ceebf@echostorm.net>
I tried this morning since it had been a few hours since the last
attempt was made and I saw the same error. Changing the SDP name didn't
have any effect on it either. I need to debug it to provide more data
but it looks to be a little problematic right now.
- Brian
On Jul 5, 2006, at 2:15 AM, Ross Finlayson wrote:
> At 08:56 PM 7/4/2006, you wrote:
>> I tried the MPEG4 darwin injector test program to see how it would
>> work
>> against my DSS server and I didn't have much luck getting it to work.
>> When live issues the PLAY command DSS returns an error stating a
>> precondition failed. I've using the latest live release and DSS 5.5.
>>
>> Can someone verify this is supposed to work in the recent live
>> releases?
>
> I don't know of any problems. Note that the "precondition failed"
> error is Darwin bogosity. It seems to occur if you try to inject a
> new stream - with the same SDP name as before - too soon after the
> last time you tried. Wait a minute or so (for Darwin to time out
> some state) and try again.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
From finlayson at live555.com Wed Jul 5 05:45:07 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 05 Jul 2006 05:45:07 -0700
Subject: [Live-devel] [PATCH]MPEG1-Audio for Kasenna
In-Reply-To: <44AA8864.70509@lincor.com>
References: <44AA8864.70509@lincor.com>
Message-ID: <7.0.1.0.1.20060705052117.01be3258@live555.com>
Grumble......
First, regarding the patch to "RTSPClient.cpp", please change it so
that you are not simply duplicating the assignment to "sdpFmt" in the
two branches. Since the only change between the two branches is
"video" vs "audio" in the "m=" line, you should be able to change
this so that it continues to assign "sdpFmt" only once.
Second, there is (to my knowledge) nothing wrong with
"StreamParser". Note that it is - as its name implies - meant for
parsing a *byte stream* source (from which you can read
arbitrarily-sized chunks of data). The problem is that you are not
feeding it a byte stream source. Instead, you are trying to feed it
a "BasicUDPSource", which "StreamParser" was not designed to read from.
Since your input (raw-UDP) packets contain MPEG audio frames as is,
then I don't see why you need any 'framer' object in front of the
"BasicUDPSource". Instead, just deliver the data, as is, to
VLC. What's that you say? VLC needs a presentation time? Gee, if
only there were some standard streaming media protocol that includes
a presentation timestamp - then Kasenna could have used that! Excuse
my sarcasm, but I hope you can see why I have such disdain for
Kasenna (a company which, by the way, I am in the process of putting
out of business).
If VLC can't handle playing MPEG audio data without a presentation
timestamp (but it should be able to, because it's able to play HTTP
MPEG audio streams just fine!), then I suggest writing a new
"MPEG1or2AudioStreamDiscreteFramer" (a subclass of "FramedFilter")
that - unlike "MPEG1or2AudioStreamFramer" - assumes that its input
data will be (integral multiples of) discrete MPEG audio
frames. (Don't tell me that Kasenna delivers its MPEG audio data so
that UDP packet boundaries don't align with MPEG audio frame
boundaries - then I'll really puke :-)
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From ncornejo at gmail.com Wed Jul 5 08:41:45 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Wed, 5 Jul 2006 09:41:45 -0600
Subject: [Live-devel] TS Files and Sound
Message-ID: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
Hello,
I'm using liveMedia libraries. I have successfully sent a TransportStream
(*.ts files) to Aminet 110 STB's. Everything is perfect, except that there
is no sound. Do you have any idea of why this may be? Other streaming
clients (like VLC) are able to play sound.
Is there a way to convert, on the fly, and MPEG2 video to a Transport
Stream??
thank you very mch for your help.
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From finlayson at live555.com Wed Jul 5 08:48:16 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 05 Jul 2006 08:48:16 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.co
m>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060705084426.01be3258@live555.com>
>I'm using liveMedia libraries. I have successfully sent a
>TransportStream (*.ts files) to Aminet 110 STB's. Everything is
>perfect, except that there is no sound. Do you have any idea of why
>this may be?
It's most likely because the audio is in a format that the Amino STB
does not handle. If I recall correctly, it can handle only MPEG
layer II audio - not layer III (i.e., MP3).
> Other streaming clients (like VLC) are able to play sound.
That's because they are better than the Amino STB :-)
>Is there a way to convert, on the fly, and MPEG2 video to a Transport Stream??
Note our "testMPEG1or2ProgramToTransportStream" demo application,
which converts a MPEG Program Stream file to a Transport Stream file.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From ncornejo at gmail.com Thu Jul 6 08:16:01 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Thu, 6 Jul 2006 09:16:01 -0600
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <7.0.1.0.1.20060705084426.01be3258@live555.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
Message-ID: <194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
Hello Ross,
Thank you for your fast response. I have analyzed the TS stream produced by
the "testMPEG1or2ProgramToTransportStream" program. Apparently, the problem
is caused because there is no Audio PID present in the stream. It would
appear that the audio and video have been multiplexed together, and plays
fine in VLC, etc, but is having a problem with the Aminet 110 STB.
Is there a way to use the LiveMedia library to create a TS stream with audio
PID ?
thank you.
Napoleon E. Cornejo
On 7/5/06, Ross Finlayson wrote:
>
>
> >I'm using liveMedia libraries. I have successfully sent a
> >TransportStream (*.ts files) to Aminet 110 STB's. Everything is
> >perfect, except that there is no sound. Do you have any idea of why
> >this may be?
>
> It's most likely because the audio is in a format that the Amino STB
> does not handle. If I recall correctly, it can handle only MPEG
> layer II audio - not layer III (i.e., MP3).
>
> > Other streaming clients (like VLC) are able to play sound.
>
> That's because they are better than the Amino STB :-)
>
>
> >Is there a way to convert, on the fly, and MPEG2 video to a Transport
> Stream??
>
> Note our "testMPEG1or2ProgramToTransportStream" demo application,
> which converts a MPEG Program Stream file to a Transport Stream file.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From finlayson at live555.com Thu Jul 6 08:22:59 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 06 Jul 2006 08:22:59 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.co
m>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060706082130.02000930@live555.com>
>Thank you for your fast response. I have analyzed the TS stream
>produced by the "testMPEG1or2ProgramToTransportStream"
>program. Apparently, the problem is caused because there is no
>Audio PID present in the stream.
Please put your original MPEG Program Stream (*not* the Transport
Stream) file on a web server, and send us the URL, so we can take a
look at this ourselves.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From ncornejo at gmail.com Thu Jul 6 09:24:27 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Thu, 6 Jul 2006 10:24:27 -0600
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <7.0.1.0.1.20060706082130.02000930@live555.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
Message-ID: <194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
Thank you, Ross.
I have placed the original mpeg video here:
http://208.131.130.36/zyban_b.mpg
And the converted to TS using liveMedia is here:
http://208.131.130.36/zyban_b.ts
My code to convert from mpeg to ts is basically this:
FramedSource* inputSource = ByteStreamFileSource::createNew(*env,
inputFile);
MPEG1or2Demux* baseDemultiplexor = MPEG1or2Demux::createNew(*env,
inputSource);
MPEG1or2DemuxedElementaryStream* pesSource =
baseDemultiplexor->newRawPESStream();
FramedSource* tsFrames = MPEG2TransportStreamFromPESSource::createNew(*env,
pesSource);
MediaSink* outputSink = FileSink::createNew(*env, outputFile);
outputSink->startPlaying(*tsFrames, afterConverting, NULL);
My code, to stream it over the network is this:
ServerMediaSession* sms = ServerMediaSession::createNew(*env, streamName,
streamName, descriptionString);
sms->addSubsession(MPEG2TransportFileServerMediaSubsession::createNew(*env,
inputFileName, False));
rtspServer->addServerMediaSession(sms);
On 7/6/06, Ross Finlayson wrote:
>
>
> >Thank you for your fast response. I have analyzed the TS stream
> >produced by the "testMPEG1or2ProgramToTransportStream"
> >program. Apparently, the problem is caused because there is no
> >Audio PID present in the stream.
>
> Please put your original MPEG Program Stream (*not* the Transport
> Stream) file on a web server, and send us the URL, so we can take a
> look at this ourselves.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From jiri.pinkava at vscht.cz Thu Jul 6 13:21:02 2006
From: jiri.pinkava at vscht.cz (pinky)
Date: Thu, 06 Jul 2006 22:21:02 +0200
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
Message-ID: <44AD70AE.5090003@vscht.cz>
There are new release of my TS Framer/Simple Framer. They are tested and
working and broken TS stream, but more testing is willcomen and necessary.
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From ncornejo at gmail.com Thu Jul 6 16:34:16 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Thu, 6 Jul 2006 16:34:16 -0700
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <194164850607061410k64a690c2l9066d74abfa2f976@mail.gmail.com>
References: <44AD70AE.5090003@vscht.cz>
<194164850607061410k64a690c2l9066d74abfa2f976@mail.gmail.com>
Message-ID: <194164850607061634r70714417r408424916a94b96e@mail.gmail.com>
Does this class add a PID for sound frames?
I was having a problem with the Aminet 110 STB with sound. The streamed
video played ok, but without sound, apparently because the sound had no PID.
The way I was doing was using the MPEG2TransportStreamFromPESSource class.
Napoleon.
> On 7/6/06, pinky wrote:
>
> > There are new release of my TS Framer/Simple Framer. They are tested and
> working and broken TS stream, but more testing is willcomen and
> necessary.
>
>
> // A filter that passes through (unchanged) chunks that contain an
> integral number
> // of MPEG-2 Transport Stream packets, but returning (in
> "fDurationInMicroseconds")
> // an updated estimate of the time gap between chunks.
> // C++ header
>
> #ifndef _MPEG2_TRANSPORT_STREAM_FRAMER_SIMPLE_HH
> #define _MPEG2_TRANSPORT_STREAM_FRAMER_SIMPLE_HH
>
> #ifndef _FRAMED_FILTER_HH
> #include "FramedFilter.hh"
> #endif
>
> #define TRANSPORT_PACKET_SIZE 188
>
> class MPEG2TransportStreamFramerSimple: public FramedFilter {
> public:
>
........ MESSAGE LENGTH TRIMMED ......
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From hartman at videolan.org Thu Jul 6 17:01:45 2006
From: hartman at videolan.org (Derk-Jan Hartman)
Date: Fri, 7 Jul 2006 02:01:45 +0200
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
Message-ID: <258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
Actually more likely the issue lies here:
main debug: `http://208.131.130.36/zyban_b.ts' successfully opened
ts debug: PATCallBack called
ts debug: new PAT ts_id=1 version=1 current_next=1
ts debug: * number=1 pid=16
ts debug: PMTCallBack called
ts debug: new PMT program number=1 version=1 pid_pcr=224
ts debug: * es pid=224 type=1 fcc=mpgv
ts debug: PMTCallBack called
ts debug: new PMT program number=1 version=2 pid_pcr=224
ts debug: * es pid=192 type=3 fcc=mpga
ts debug: * es pid=224 type=1 fcc=mpgv
liveMedia sends a PMT with only the video track, and then (when in
the PS it finds another track) sends an updated PMT which contains
all tracks. The amino probably does not support updating of the PMT
table. That's a big bug btw, and you should contact them to get that
fixed.
DJ
On 6-jul-2006, at 18:24, Napoleon Cornejo wrote:
> Thank you, Ross.
>
> I have placed the original mpeg video here:
> http://208.131.130.36/zyban_b.mpg
>
> And the converted to TS using liveMedia is here:
> http://208.131.130.36/zyban_b.ts
>
> My code to convert from mpeg to ts is basically this:
>
> FramedSource* inputSource = ByteStreamFileSource::createNew(*env,
> inputFile);
> MPEG1or2Demux* baseDemultiplexor = MPEG1or2Demux::createNew(*env,
> inputSource);
> MPEG1or2DemuxedElementaryStream* pesSource = baseDemultiplexor-
> >newRawPESStream();
> FramedSource* tsFrames =
> MPEG2TransportStreamFromPESSource::createNew(*env, pesSource);
> MediaSink* outputSink = FileSink::createNew(*env, outputFile);
> outputSink->startPlaying(*tsFrames, afterConverting, NULL);
>
> My code, to stream it over the network is this:
>
> ServerMediaSession* sms = ServerMediaSession::createNew(*env,
> streamName, streamName, descriptionString);
> sms->addSubsession
> (MPEG2TransportFileServerMediaSubsession::createNew(*env,
> inputFileName, False));
> rtspServer->addServerMediaSession(sms);
>
>
>
>
> On 7/6/06, Ross Finlayson < finlayson at live555.com> wrote:
> >Thank you for your fast response. I have analyzed the TS stream
> >produced by the "testMPEG1or2ProgramToTransportStream"
> >program. Apparently, the problem is caused because there is no
> >Audio PID present in the stream.
>
> Please put your original MPEG Program Stream (*not* the Transport
> Stream) file on a web server, and send us the URL, so we can take a
> look at this ourselves.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
> < http://www.live555.com/>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
> --
> Napoleon E. Cornejo
> ncornejo at gmail.com
> San Salvador, 2006
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
From hartman at videolan.org Thu Jul 6 18:02:05 2006
From: hartman at videolan.org (Derk-Jan Hartman)
Date: Fri, 7 Jul 2006 03:02:05 +0200
Subject: [Live-devel] [PATCH]MPEG1-Audio for Kasenna
In-Reply-To: <7.0.1.0.1.20060705052117.01be3258@live555.com>
References: <44AA8864.70509@lincor.com>
<7.0.1.0.1.20060705052117.01be3258@live555.com>
Message-ID: <1E3C05F4-28DB-4A69-9157-8C4CF2F48B06@videolan.org>
On 5-jul-2006, at 14:45, Ross Finlayson wrote:
> Grumble......
>
> First, regarding the patch to "RTSPClient.cpp", please change it so
> that you are not simply duplicating the assignment to "sdpFmt" in the
> two branches. Since the only change between the two branches is
> "video" vs "audio" in the "m=" line, you should be able to change
> this so that it continues to assign "sdpFmt" only once.
>
> Second, there is (to my knowledge) nothing wrong with
> "StreamParser". Note that it is - as its name implies - meant for
> parsing a *byte stream* source (from which you can read
> arbitrarily-sized chunks of data). The problem is that you are not
> feeding it a byte stream source. Instead, you are trying to feed it
> a "BasicUDPSource", which "StreamParser" was not designed to read
> from.
>
> Since your input (raw-UDP) packets contain MPEG audio frames as is,
> then I don't see why you need any 'framer' object in front of the
> "BasicUDPSource". Instead, just deliver the data, as is, to
> VLC. What's that you say? VLC needs a presentation time? Gee, if
> only there were some standard streaming media protocol that includes
> a presentation timestamp - then Kasenna could have used that! Excuse
> my sarcasm, but I hope you can see why I have such disdain for
> Kasenna (a company which, by the way, I am in the process of putting
> out of business).
They don't need you for that. The badmouthing everyone does about
them should take care of it eventually.
I have not ever spoken to a satisfied Kasenna customer, and that
includes Glen Gray :D
Unfortunately not that many server platforms do Trickplay.
> If VLC can't handle playing MPEG audio data without a presentation
> timestamp (but it should be able to, because it's able to play HTTP
> MPEG audio streams just fine!), then I suggest writing a new
> "MPEG1or2AudioStreamDiscreteFramer" (a subclass of "FramedFilter")
> that - unlike "MPEG1or2AudioStreamFramer" - assumes that its input
> data will be (integral multiples of) discrete MPEG audio
> frames. (Don't tell me that Kasenna delivers its MPEG audio data so
> that UDP packet boundaries don't align with MPEG audio frame
> boundaries - then I'll really puke :-)
Force VLC to use the mpga demuxer (much like the TS demux is forced)
then it will play. I'm not sure if seeking and pausing is still
possible then however (I seriously doubt it). Perhaps simply
specifying the data to be "unpacketized" mpga/mp3 data will work as
well. In that case the packetizer should kick in to clean up the
data, timestamp it etc.
DJ
From finlayson at live555.com Thu Jul 6 18:19:07 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 06 Jul 2006 18:19:07 -0700
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <44AD70AE.5090003@vscht.cz>
References: <44AD70AE.5090003@vscht.cz>
Message-ID: <7.0.1.0.1.20060706181637.01c90860@live555.com>
At 01:21 PM 7/6/2006, you wrote:
>There are new release of my TS Framer/Simple Framer.
Rather than simply sending code for this new class (which I have no
plans to add to the library) without explanation, it might be more
instructive if you explained what you think is wrong with the
existing class, and what your new class does that supposedly improves upon it.
If there really is a problem with the existing code, then the best
solution would be to fix that, because that's what everyone is going
to get to see.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Fri Jul 7 21:24:46 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 07 Jul 2006 21:24:46 -0700
Subject: [Live-devel] Darwin Injector
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060707212320.02035fb0@live555.com>
At 08:56 PM 7/4/2006, you wrote:
>I tried the MPEG4 darwin injector test program to see how it would work
>against my DSS server and I didn't have much luck getting it to work.
>When live issues the PLAY command DSS returns an error stating a
>precondition failed. I've using the latest live release and DSS 5.5.
>
>Can someone verify this is supposed to work in the recent live releases?
FYI, I verified today that the "testMPEG4VideoToDarwin" demo
application (which uses "DarwinInjector") still works OK (with DSS 5.5).
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Fri Jul 7 21:31:30 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 07 Jul 2006 21:31:30 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
Message-ID: <7.0.1.0.1.20060707212503.01ffec80@live555.com>
FYI, I verified today that Napoleon Cornejo's "zyban_b.ts" file does
- as he reported - play on the Amino STB (when streamed) without
audio. It's not clear to me what the problem might be. However, I
doubt that it has anything to do with PIDs, PMTs or PATs, because the file
http://www.live555.com/test.ts
which was also generated in the same way (in this case, by the
"testMPEG1or2ProgramToTransportStream" demo application), and which
makes similar use of PIDs, PMTs or PATs - plays OK, with audio as
well as video.
Something - it's not clear what - about the audio in the "zyban_b.ts"
file (and the original Program Stream file) is making the Amino STB
unhappy. I suggest reporting this problem to Amino; perhaps they
will know what the problem is.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From micraltoo at gmail.com Sun Jul 9 08:07:14 2006
From: micraltoo at gmail.com (Bus Mini)
Date: Sun, 9 Jul 2006 23:07:14 +0800
Subject: [Live-devel] How large and how fast can a file trevels via LAN uing
this library?
Message-ID:
Hi,
I am new in this list. Before I start to use this library, I have two
questions to asked.
1, can I send othet data tpye(ect. jpg) using this library? It seems
there is no API for me to do this.
2, How fast and how large can a file trevels via LAN uing this
library? Because I know some libraries have this constrain. If the
data was sended fast then a library can deal with, it will cause
flowup.
Thanks for reanding my mail.
--
Yours sincerely
Mini Bus.
From liufei at zzvcom.com Sun Jul 9 23:07:23 2006
From: liufei at zzvcom.com (liufei at zzvcom.com)
Date: Mon, 10 Jul 2006 14:07:23 +0800
Subject: [Live-devel] How to get the traffic statistics of one
RTSPClientSession?
Message-ID: <44B1EE9B.000009.45044@0TQNTF8WIHSH>
How to get the traffic statistics of one RTSPClientSession?
Would I insert some code in class RTCPInstance?
But how can I get the client's id in RTCPInstance?
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From xinzhang.cao at gmail.com Mon Jul 10 00:10:21 2006
From: xinzhang.cao at gmail.com (xinzhang cao)
Date: Mon, 10 Jul 2006 15:10:21 +0800
Subject: [Live-devel] How to get the traffic statistics of one
RTSPClientSession?
In-Reply-To: <44B1EE9B.000009.45044@0TQNTF8WIHSH>
References: <44B1EE9B.000009.45044@0TQNTF8WIHSH>
Message-ID: <48f487430607100010v28d8779aicda9e3b04ab22105@mail.gmail.com>
??????????
?????????RTCPInstance????????????????
?06-7-10?liufei at zzvcom.com ???
>
> How to get the traffic statistics of one RTSPClientSession?
> Would I insert some code in class RTCPInstance?
> But how can I get the client's id in RTCPInstance?
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From yossyd at nayos.com Mon Jul 10 04:00:31 2006
From: yossyd at nayos.com (Yossy Dreyfus)
Date: Mon, 10 Jul 2006 13:00:31 +0200
Subject: [Live-devel] LIVE555 and "C"
Message-ID: <000601c6a410$12fe1030$570a1f0a@nayos.local>
I need to use the classes in a "C" code. What is the best way to do
that?
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From micraltoo at gmail.com Mon Jul 10 06:45:01 2006
From: micraltoo at gmail.com (Bus Mini)
Date: Mon, 10 Jul 2006 21:45:01 +0800
Subject: [Live-devel] How large and how fast can a file trevels via LAN
uing this library?
In-Reply-To:
References:
Message-ID:
Could somebody answer my question, because this library so large, it
is diffcult for me to find the answers. I need your help.
Thanks for reading my mail.
On 7/9/06, Bus Mini wrote:
> Hi,
>
> I am new in this list. Before I start to use this library, I have two
> questions to asked.
>
> 1, can I send othet data tpye(ect. jpg) using this library? It seems
> there is no API for me to do this.
>
> 2, How fast and how large can a file trevels via LAN uing this
> library? Because I know some libraries have this constrain. If the
> data was sended fast then a library can deal with, it will cause
> flowup.
>
> Thanks for reanding my mail.
>
> --
> Yours sincerely
> Mini Bus.
>
--
Yours sincerely
Mini Bus.
From mbernal at dif.um.es Mon Jul 10 09:56:34 2006
From: mbernal at dif.um.es (Manuel Bernal Llinares)
Date: Mon, 10 Jul 2006 18:56:34 +0200
Subject: [Live-devel] How large and how fast can a file trevels via LAN
uing this library?
In-Reply-To:
References:
Message-ID: <200607101856.35116.mbernal@dif.um.es>
El Lunes, 10 de Julio de 2006 15:45, Bus Mini escribi?:
> Could somebody answer my question, because this library so large, it
> is diffcult for me to find the answers. I need your help.
> Thanks for reading my mail.
>
> On 7/9/06, Bus Mini wrote:
> > Hi,
> >
> > I am new in this list. Before I start to use this library, I have two
> > questions to asked.
> >
> > 1, can I send othet data tpye(ect. jpg) using this library? It seems
> > there is no API for me to do this.
To do this you'll need to implement a class hierarchy that could understand
JPEG images on the server side to send them over the net and, on the client
side, the right RTP Source to get them from the net. It is saying, as I've
seen, this library has two hierarhies of classes: one for the server and
related to Sinks, and one for the client and related to Sources. Look at
classes like MPEG1or2VideoRTPSink and MPEG1or2VideoSource, the first is used
by the server and the second is used by the client.
> >
> > 2, How fast and how large can a file trevels via LAN uing this
> > library? Because I know some libraries have this constrain. If the
> > data was sended fast then a library can deal with, it will cause
> > flowup.
I've been working with this library several months to add some features like
IPv6 and, although I don't know the whole library because I haven't had to
modify the whole library, it seems that the throughput this software supports
depends on the Sinks hierarchy of classes which are the responsible of
injecting network traffic.
This traffic depends on the file, it is saying, it depends on whether you are
sending mp3 music, mpeg audio/video and so on, I think.
So, taking a hard source of information like a .VOB file would be a good
example of the capabilities the library has to inject traffic on the net.
But as my work with this liveMedia library is related to security and IP
protocol support, I don't know much about the classes that decode media files
for sending over the net.
We must also keep in mind that, in most cases, it will be UDP traffic, which
will be under the restrictions of the net the library is working.
I hope this could be of some kind of help to you.
> >
> > Thanks for reanding my mail.
> >
> > --
> > Yours sincerely
> > Mini Bus.
--
Manuel Bernal Llinares
Dept. Ingenier?a de la Informaci?n y las Comunicaciones
Facultad de Inform?tica
Universidad de Murcia
Laboratorio 1.02 (c?digo 009)
30001 Murcia
Extensi?n: 4644
Tlf.: (+34) 655 703 911
Gloffing is a state of mine.
From ncornejo at gmail.com Mon Jul 10 13:45:16 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Mon, 10 Jul 2006 14:45:16 -0600
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <7.0.1.0.1.20060707212503.01ffec80@live555.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
Message-ID: <194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
I don't know if you tried converting from the original mpeg file, but when I
did, errors like the following appeared:
MPEGProgramStreamParser::parsePESPacket() error: PES_packet_length (2036)
exceed s max frame size asked for (220)
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Missing sync byte!
Probably this error is generating a messy transport stream that the Amino
PLayer is unable to play properly. Could this be possible?
Napoleon.
On 7/7/06, Ross Finlayson wrote:
>
> FYI, I verified today that Napoleon Cornejo's "zyban_b.ts" file does
> - as he reported - play on the Amino STB (when streamed) without
> audio. It's not clear to me what the problem might be. However, I
> doubt that it has anything to do with PIDs, PMTs or PATs, because the file
> http://www.live555.com/test.ts
> which was also generated in the same way (in this case, by the
> "testMPEG1or2ProgramToTransportStream" demo application), and which
> makes similar use of PIDs, PMTs or PATs - plays OK, with audio as
> well as video.
>
> Something - it's not clear what - about the audio in the "zyban_b.ts"
> file (and the original Program Stream file) is making the Amino STB
> unhappy. I suggest reporting this problem to Amino; perhaps they
> will know what the problem is.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From micraltoo at gmail.com Tue Jul 11 06:54:13 2006
From: micraltoo at gmail.com (Bus Mini)
Date: Tue, 11 Jul 2006 21:54:13 +0800
Subject: [Live-devel] How can I get them all right?
Message-ID:
Hi all,
I am sorry to trouble you, I tried to install the library code onto my
computer.After I installing, I tried to compile a test program, compiler
told me it can not find "liveMedia.hh", so, I changed its path to
"/...../liveMedia.hh", but then , in another head file it can not find
another head file(ect: include"NetCommon.h"). I guess maybe Imy installing
was failed. But during my installing, I didn't find any exception.
I just run "./genMakefiles macosx" then run "make". Could anybody hlep me?
It is very kind of you to do that.
Thanks for reading my mail.
--
Yours sincerely
Mini Bus.
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From ncornejo at gmail.com Tue Jul 11 07:37:32 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Tue, 11 Jul 2006 07:37:32 -0700
Subject: [Live-devel] How can I get them all right?
In-Reply-To:
References:
Message-ID: <194164850607110737q54021b9bx8be7f901bfed24e9@mail.gmail.com>
I'm not sure how MacOS works, but probably check your environment variable
pointing to your include file folder.
On 7/11/06, Bus Mini wrote:
>
> Hi all,
> I am sorry to trouble you, I tried to install the library code onto my
> computer.After I installing, I tried to compile a test program, compiler
> told me it can not find "liveMedia.hh", so, I changed its path to
> "/...../liveMedia.hh", but then , in another head file it can not find
> another head file(ect: include" NetCommon.h"). I guess maybe Imy
> installing was failed. But during my installing, I didn't find any
> exception.
> I just run "./genMakefiles macosx" then run "make". Could anybody hlep
> me? It is very kind of you to do that.
> Thanks for reading my mail.
> --
> Yours sincerely
> Mini Bus.
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From finlayson at live555.com Tue Jul 11 09:24:02 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 11 Jul 2006 09:24:02 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com
>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
<194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060711092055.01f6d378@live555.com>
At 01:45 PM 7/10/2006, you wrote:
>I don't know if you tried converting from the original mpeg file,
>but when I did, errors like the following appeared:
>
>MPEGProgramStreamParser::parsePESPacket() error: PES_packet_length
>(2036) exceed s max frame size asked for (220)
>Missing sync byte!
>Missing sync byte!
>Missing sync byte!
When I converted your Program Stream file to a Transport Stream using
the "testMPEG1or2ProgramToTransportStream" demo application, I saw no
such errors. If you didn't use that code, then I can't help you - sorry.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From ncornejo at gmail.com Tue Jul 11 13:47:35 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Tue, 11 Jul 2006 13:47:35 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <7.0.1.0.1.20060711092055.01f6d378@live555.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
<194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
<7.0.1.0.1.20060711092055.01f6d378@live555.com>
Message-ID: <194164850607111347y78687fbfn8ecf7fa938ff4197@mail.gmail.com>
Thank you, Ross. The code that I used to convert it is basically a
copy-paste from the code of "testMPEG1or2ProgramToTransportStream"
FramedSource* inputSource = ByteStreamFileSource::createNew(*env,
inputFile);
MPEG1or2Demux* baseDemultiplexor = MPEG1or2Demux::createNew(*env,
inputSource);
MPEG1or2DemuxedElementaryStream* pesSource =
baseDemultiplexor->newRawPESStream();
FramedSource* tsFrames = MPEG2TransportStreamFromPESSource::createNew(*env,
pesSource);
MediaSink* outputSink = FileSink::createNew(*env, outputFile);
outputSink->startPlaying(*tsFrames, afterConverting, NULL);
perhaps there is something wrong with my use of the code. Could you provide
me the source program stream of http://www.live555.com/test.ts so that I can
try converting it please? I'll check to see if those errors I mentioned
before show up again.
Napoleon.
On 7/11/06, Ross Finlayson wrote:
>
> At 01:45 PM 7/10/2006, you wrote:
> >I don't know if you tried converting from the original mpeg file,
> >but when I did, errors like the following appeared:
> >
> >MPEGProgramStreamParser::parsePESPacket() error: PES_packet_length
> >(2036) exceed s max frame size asked for (220)
> >Missing sync byte!
> >Missing sync byte!
> >Missing sync byte!
>
> When I converted your Program Stream file to a Transport Stream using
> the "testMPEG1or2ProgramToTransportStream" demo application, I saw no
> such errors. If you didn't use that code, then I can't help you - sorry.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From finlayson at live555.com Tue Jul 11 14:25:54 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 11 Jul 2006 14:25:54 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607111347y78687fbfn8ecf7fa938ff4197@mail.gmail.co
m>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<7.0.1.0.1.20060705084426.01be3258@live555.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
<194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
<7.0.1.0.1.20060711092055.01f6d378@live555.com>
<194164850607111347y78687fbfn8ecf7fa938ff4197@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060711142502.01f93848@live555.com>
>Could you provide me the source program stream of
>http://www.live555.com/test.ts so
>that I can try converting it please?
OK, it's at
http://www.live555.com/test.mpg
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From sdhays.neon.com.tw at gmail.com Tue Jul 11 17:53:16 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Wed, 12 Jul 2006 08:53:16 +0800
Subject: [Live-devel] How can I get them all right?
In-Reply-To: <194164850607110737q54021b9bx8be7f901bfed24e9@mail.gmail.com>
References:
<194164850607110737q54021b9bx8be7f901bfed24e9@mail.gmail.com>
Message-ID: <9866ce4f0607111753q694e470m75d3d1f09dd6856b@mail.gmail.com>
Where did you install the library? The 'make' command only compiles the
library and its test programs. You have to manually install it (if you want
to) and ensure that your library search path includes the live/liveMedia,
live/UsageEnvironment, live/BasicUsageEnvironment, and live/groupsock
directories and the "include" sub-directories of those directories are on
your header search path.
Scott
On 7/11/06, Napoleon Cornejo wrote:
>
> I'm not sure how MacOS works, but probably check your environment variable
> pointing to your include file folder.
>
> On 7/11/06, Bus Mini < micraltoo at gmail.com> wrote:
>
> > Hi all,
> > I am sorry to trouble you, I tried to install the library code onto my
> > computer.After I installing, I tried to compile a test program, compiler
> > told me it can not find "liveMedia.hh", so, I changed its path to
> > "/...../liveMedia.hh", but then , in another head file it can not find
> > another head file(ect: include" NetCommon.h"). I guess maybe Imy
> > installing was failed. But during my installing, I didn't find any
> > exception.
> > I just run "./genMakefiles macosx" then run "make". Could anybody hlep
> > me? It is very kind of you to do that.
> > Thanks for reading my mail.
> > --
> > Yours sincerely
> > Mini Bus.
> >
> > _______________________________________________
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> >
> >
> >
>
>
> --
> Napoleon E. Cornejo
> ncornejo at gmail.com
> San Salvador, 2006
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From yoann.ramard at orange-ft.com Wed Jul 12 08:21:51 2006
From: yoann.ramard at orange-ft.com (zze-RAMARD Yoann RD-TECH-GRE)
Date: Wed, 12 Jul 2006 17:21:51 +0200
Subject: [Live-devel] Streaming from one source to multiple sink
Message-ID:
Hi,
My problem is simple : I do not want to use multicast streaming. I have
a unicast source that is streamed to a relay and I want to relay it to
multiple destinations using unicast (because of an eventual transcode
for some destination). One unicast sink&socket for one destination. If I
try to play directly the source for all the sinks, I get the normal
error that I attemp to read more than once at the same time. How can I
do if it is possible to?
Yoann
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From mbernal at dif.um.es Wed Jul 12 08:44:56 2006
From: mbernal at dif.um.es (Manuel Bernal Llinares)
Date: Wed, 12 Jul 2006 17:44:56 +0200
Subject: [Live-devel] Streaming from one source to multiple sink
In-Reply-To:
References:
Message-ID: <200607121744.56696.mbernal@dif.um.es>
El Mi?rcoles, 12 de Julio de 2006 17:21, zze-RAMARD Yoann RD-TECH-GRE
escribi?:
> Hi,
> My problem is simple : I do not want to use multicast streaming. I have
> a unicast source that is streamed to a relay and I want to relay it to
> multiple destinations using unicast (because of an eventual transcode
> for some destination). One unicast sink&socket for one destination. If I
> try to play directly the source for all the sinks, I get the normal
> error that I attemp to read more than once at the same time. How can I
> do if it is possible to?
> Yoann
Uuuumm, this is quite interesting, I think you should place something like a
proxy between the source (which is only one) and the sinks (which are many).
This way, your proxy would read each frame from the source and would replay
this frame over each sink. This could be an initial approach.
--
Manuel Bernal Llinares
Dept. Ingenier?a de la Informaci?n y las Comunicaciones
Facultad de Inform?tica
Universidad de Murcia
Laboratorio 1.02 (c?digo 009)
30001 Murcia
Extensi?n: 4644
Tlf.: (+34) 655 703 911
"Everyone is entitled to an *informed* opinion."
-- Harlan Ellison
From sdhays.neon.com.tw at gmail.com Wed Jul 12 18:19:06 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Thu, 13 Jul 2006 09:19:06 +0800
Subject: [Live-devel] Streaming from one source to multiple sink
In-Reply-To:
References:
Message-ID: <9866ce4f0607121819j5bfbc652pb8cca4682dc4f422@mail.gmail.com>
Are you using reuseFirstSource=True?
Scott
On 7/12/06, zze-RAMARD Yoann RD-TECH-GRE wrote:
>
> Hi,
> My problem is simple : I do not want to use multicast streaming. I have a
> unicast source that is streamed to a relay and I want to relay it to
> multiple destinations using unicast (because of an eventual transcode for
> some destination). One unicast sink&socket for one destination. If I try to
> play directly the source for all the sinks, I get the normal error that I
> attemp to read more than once at the same time. How can I do if it is
> possible to?
>
> Yoann
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From tjc103 at ecs.soton.ac.uk Thu Jul 13 01:08:58 2006
From: tjc103 at ecs.soton.ac.uk (tjc103 at ecs.soton.ac.uk)
Date: Thu, 13 Jul 2006 09:08:58 +0100
Subject: [Live-devel] nice -1 required on server side
Message-ID: <1152778138.44b5ff9a476f7@webmail.soton.ac.uk>
Hi,
I've been looking at the live555 libraries for a research project at uni and
have managed to get all the test programs working that come with the
library.
At the moment I send with testMPEG2TransportStreamer (compiled with
IMPLEMENT_RTSP_SERVER defined and I only use unicast) and on my other box I
receive with openRTSP, which writes to a file.
Playing back this file however with VLC shows several glitches. These can be
overcome though if I execute testMPEG2TransportStreamer like this :
$ nice -1 ./testMPEG2TransportStreamer
This gives fault less playback on a 96 second clip looping all night. I was
just wondering if anyone else has experienced this? Is this normal, or is
there anyway of overcoming it?
I've also tried streaming a TS with VLC, however by default it seems to
demux the video stream when executed like this:
$ vlc -vvv some.ts --sout
'#rtp{dst=192.168.0.254,port=1234,sdp=rtsp://192.168.0.1:8080/test.sdp}'
Adding ,mux=ts before the closing braket causes openRTSP to not find the
stream at all. I know this isn't a VLC list, but I'm sure someone here must
of tested a client on VLC at some point. Any idea's what I'm doing wrong?
Thanks
Theo
From jiri.pinkava at vscht.cz Thu Jul 13 02:12:46 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Thu, 13 Jul 2006 11:12:46 +0200
Subject: [Live-devel] nice -1 required on server side
In-Reply-To: <1152778138.44b5ff9a476f7@webmail.soton.ac.uk>
References: <1152778138.44b5ff9a476f7@webmail.soton.ac.uk>
Message-ID: <20060713111246.38684964.jiri.pinkava@vscht.cz>
On Thu, 13 Jul 2006 09:08:58 +0100
tjc103 at ecs.soton.ac.uk wrote:
> Hi,
>
> I've been looking at the live555 libraries for a research project at uni and
> have managed to get all the test programs working that come with the
> library.
>
> At the moment I send with testMPEG2TransportStreamer (compiled with
> IMPLEMENT_RTSP_SERVER defined and I only use unicast) and on my other box I
> receive with openRTSP, which writes to a file.
If Your goal is stream only this short clip (or something else from file), look for testMPEG2AudioVideoStreamer (You may use vlc for conversion the clip from TS stream to MPEG2 Program Stream).
>
> Playing back this file however with VLC shows several glitches. These can be
> overcome though if I execute testMPEG2TransportStreamer like this :
>
> $ nice -1 ./testMPEG2TransportStreamer
>
> This gives fault less playback on a 96 second clip looping all night. I was
> just wondering if anyone else has experienced this? Is this normal, or is
> there anyway of overcoming it?
This is not normal, at first look at the CPU load (is there no other process which take processor time?). I have some problems with TS streaming (for example CPU load and timing), but I use DVB-T card as source. This is probably caused by MPEG2TransportStreamFramer class, You may try change some parameters in this class (in .cpp file at top) but this probably does not fully solve this problem. If You use file as video source, there is no other (simple) way to solve this).
>
> I've also tried streaming a TS with VLC, however by default it seems to
> demux the video stream when executed like this:
>
> $ vlc -vvv some.ts --sout
> '#rtp{dst=192.168.0.254,port=1234,sdp=rtsp://192.168.0.1:8080/test.sdp}'
>
> Adding ,mux=ts before the closing braket causes openRTSP to not find the
> stream at all. I know this isn't a VLC list, but I'm sure someone here must
> of tested a client on VLC at some point. Any idea's what I'm doing wrong?
>
Before long, long time i use VLC to stream MPEG TS. I simply copy an example form videolan.org documentation site.
> Thanks
>
> Theo
>
>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
From yoann.ramard at orange-ft.com Thu Jul 13 04:11:51 2006
From: yoann.ramard at orange-ft.com (zze-RAMARD Yoann RD-TECH-GRE)
Date: Thu, 13 Jul 2006 13:11:51 +0200
Subject: [Live-devel] Streaming from one source to multiple sink
Message-ID:
Actually, I'm not using a FileServerMediaSubsession. In fact, I'm using
testMP3Streamer without SSM or RTSPServer to stream unicast via RTP to
the IP:port of the computer where my relay application is running. The
relay gets this stream and I want to diffuse it simultaneously to
multiple players on other computers (IP:port) using the same unicast way
(no SSM or RTSPServer). Is it possible? Thanks for your answers.
Yoann
>Are you using reuseFirstSource=True?
>
>Scott
>
>On 7/12/06, zze-RAMARD Yoann RD-TECH-GRE wrote:
>>
>> Hi,
>> My problem is simple : I do not want to use multicast streaming. I
have a
>> unicast source that is streamed to a relay and I want to relay it to
>> multiple destinations using unicast (because of an eventual transcode
for
>> some destination). One unicast sink&socket for one destination. If I
try to
>> play directly the source for all the sinks, I get the normal error
that I
>> attemp to read more than once at the same time. How can I do if it is
>> possible to?
>>
>> Yoann
>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>>
>>
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From bidibulle at operamail.com Thu Jul 13 04:28:59 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 13 Jul 2006 12:28:59 +0100
Subject: [Live-devel] Streaming from one source to multiple sink
Message-ID: <20060713112859.ACE8B7B5F2@ws5-10.us4.outblaze.com>
Yoann,
see http://lists.live555.com/pipermail/live-devel/2005-December/003673.html
David
> ----- Original Message -----
> From: "zze-RAMARD Yoann RD-TECH-GRE"
> To: live-devel at ns.live555.com
> Subject: [Live-devel] Streaming from one source to multiple sink
> Date: Thu, 13 Jul 2006 13:11:51 +0200
>
>
> Actually, I'm not using a FileServerMediaSubsession. In fact, I'm using
> testMP3Streamer without SSM or RTSPServer to stream unicast via RTP to
> the IP:port of the computer where my relay application is running. The
> relay gets this stream and I want to diffuse it simultaneously to
> multiple players on other computers (IP:port) using the same unicast way
> (no SSM or RTSPServer). Is it possible? Thanks for your answers.
> Yoann
>
> > Are you using reuseFirstSource=True?
> >
> > Scott
> >
> > On 7/12/06, zze-RAMARD Yoann RD-TECH-GRE orange-ft.com> wrote:
> >>
> >> Hi,
> >> My problem is simple : I do not want to use multicast streaming. I
> have a
> >> unicast source that is streamed to a relay and I want to relay it to
> >> multiple destinations using unicast (because of an eventual transcode
> for
> >> some destination). One unicast sink&socket for one destination. If I
> try to
> >> play directly the source for all the sinks, I get the normal error
> that I
> >> attemp to read more than once at the same time. How can I do if it is
> >> possible to?
> >>
> >> Yoann
> >>
> >> _______________________________________________
> >> live-devel mailing list
> >> live-devel at lists.live555.com
> >> http://lists.live555.com/mailman/listinfo/live-devel
> >>
> >>
> >>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 9 at http://www.opera.com
Powered by Outblaze
From ibaldine at rti.org Thu Jul 13 08:35:08 2006
From: ibaldine at rti.org (Baldine, Ilia)
Date: Thu, 13 Jul 2006 11:35:08 -0400
Subject: [Live-devel] wis-streamer image freezing
Message-ID:
Just got the latest live library (07-05 i believe) and the source for
wis-streamer. Compiled the OSS WISGO driver. Used VLC to play the
stream. Things work for about 30 seconds, then the image freezes and
wis-streamer needs to be restarted (if a player is restarted, an image
shows up frozen, which suggests that wis-streamer is not totally frozen
- at least the RTSP part responds). Has anyone seen this before I dive
into the code? Does not look like the device driver issue, since simply
restarting wis-streamer seems to do the trick (for another 30 seconds).
I'm running it on RH7.3, kernel 2.6.12-something. Playing from Win XP
using VLC. Have an older proprietary WIS driver + custom RTSP streamer
(using live555) I wrote under 2.4 that seem to work fine.
-ilia
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From xcsmith at rockwellcollins.com Thu Jul 13 08:58:42 2006
From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com)
Date: Thu, 13 Jul 2006 10:58:42 -0500
Subject: [Live-devel] RTSP Documentation
Message-ID:
Hello, I've downloaded the Live555 library ("live555-latest.tar.gz") from
http://www.live555.com/liveMedia/public/
Right now I have a lot of questions about RTSP. I've been reading the RFC
drafts from rtsp.org
(http://www.rtsp.org/2003/drafts/draft05/draft-ietf-mmusic-rfc2326bis-05.txt),
among other things.
How can I find the version of the RFC that was used when writing my version
of the library? How can I find out what parts of the RTSP standard are
implemented?
Thanks!
~Medra
From tullio.chersi at tin.it Thu Jul 13 09:45:01 2006
From: tullio.chersi at tin.it (Tullio Chersi)
Date: Thu, 13 Jul 2006 18:45:01 +0200
Subject: [Live-devel] Mplayer compilation error
Message-ID: <44B6788D.5070808@tin.it>
This error mesage is the result of compiling the latest version of
Mplayer(1.0pre8) with --enable-live option. Live version is 2006.07.04.
But it compiles with an earlier version of live which I downloaded as
live-latest on 2005.08.24. Linux is SuSE 9.3 on a Pentium II Deschutes.
Gcc is 3.3.5. Mozilla 1.7.13 with mplayer plugin 3.25 works with
WindowsMedia, Quicktime and RealPlayer streams (the latter with Helix
Plugins).
libmpdemux/libmpdemux.a(demux_rtp.o)(.text+0xc2): In function
`_GLOBAL__I_rtspStreamOverTCP':
demux_rtp.cpp: undefined reference to `SECOND'
libmpdemux/libmpdemux.a(demux_rtp.o)(.text+0xfbe): In function
`demux_open_rtp':
demux_rtp.cpp: undefined reference to
`RTSPClient::setupMediaSubsession(MediaSubsession&, unsigned, unsigned)'
collect2: ld returned 1 exit status
make: *** [mplayer] Error 1
From ncornejo at gmail.com Thu Jul 13 11:07:39 2006
From: ncornejo at gmail.com (Napoleon Cornejo)
Date: Thu, 13 Jul 2006 12:07:39 -0600
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <7.0.1.0.1.20060711142502.01f93848@live555.com>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
<194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
<7.0.1.0.1.20060711092055.01f6d378@live555.com>
<194164850607111347y78687fbfn8ecf7fa938ff4197@mail.gmail.com>
<7.0.1.0.1.20060711142502.01f93848@live555.com>
Message-ID: <194164850607131107j59819f26s9b9947108c5717f0@mail.gmail.com>
Hi Ross,
I tested the test.mpg that you provided, converted it using the
testMPEG1or2ProgramToTransportStream program, and effectively the A110 STB
played it correctly with sound.
It is indeed strange why sound of other videos is not played at all. I
tried with two other mpg program sources and none played sound. They were
all converted using testMPEG1or2ProgramToTransportStream. You can find them
here:
http://208.131.130.36/coca.mpg -> http://208.131.130.36/coca.ts
http://208.131.130.36/abue.mpg -> http://208.131.130.36/abue.ts
Could it be some kind of encoding of sound with the streams I'm using?
Napoleon E. Cornejo
ncornejo at gmail.com
On 7/11/06, Ross Finlayson wrote:
>
>
> >Could you provide me the source program stream of
> >http://www.live555.com/test.ts so
> >that I can try converting it please?
>
> OK, it's at
> http://www.live555.com/test.mpg
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
Napoleon E. Cornejo
ncornejo at gmail.com
San Salvador, 2006
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From darnold at futurec.net Thu Jul 13 11:25:33 2006
From: darnold at futurec.net (David Arnold)
Date: Thu, 13 Jul 2006 11:25:33 -0700
Subject: [Live-devel] Playing streamed MP3 Audio with Quicktime
Message-ID:
Can Quicktime play MP3 audio streamed from TestOnDemandRTSPServer.cpp? I
have tried all permutations of "STREAM_USING_ADUS" and "INTERLEAVE_ADUS",
but nothing seems to work. I am able to play the audio using VLC, but
when I use QuickTime, no audio is heard and QuickTime seems to get hung. I
am using QuickTime version 7.1 on Windows XP.
Thank you,
Dave Arnold
Future Concepts, La Verne CA
The information contained in this electronic mail transmission is intended only for the use of the individual or entity named above and is privileged and confidential. If you are not the intended recipient, please do not read, copy, use or disclose this communication to others. Any dissemination, distribution or copying of this communication other than to the person or entity named above is strictly prohibited. If you have received this communication in error, please immediately delete it from your system.
From xcsmith at rockwellcollins.com Thu Jul 13 13:20:51 2006
From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com)
Date: Thu, 13 Jul 2006 15:20:51 -0500
Subject: [Live-devel] Playing streamed MP3 Audio with Quicktime
In-Reply-To:
Message-ID:
Dave,
I'm new here, but I think that problem is mentioned on the livemedia
website
http://www.live555.com/liveMedia/faq.html#quicktime-player-mp3-bug
~Medra
"David Arnold"
To
Sent by:
live-devel-bounce cc
s at ns.live555.com
Subject
[Live-devel] Playing streamed MP3
07/13/2006 01:25 Audio with Quicktime
PM
Please respond to
LIVE555 Streaming
Media -
development & use
Can Quicktime play MP3 audio streamed from TestOnDemandRTSPServer.cpp? I
have tried all permutations of "STREAM_USING_ADUS" and "INTERLEAVE_ADUS",
but nothing seems to work. I am able to play the audio using VLC, but
when I use QuickTime, no audio is heard and QuickTime seems to get hung. I
am using QuickTime version 7.1 on Windows XP.
Thank you,
Dave Arnold
Future Concepts, La Verne CA
The information contained in this electronic mail transmission is intended
only for the use of the individual or entity named above and is privileged
and confidential. If you are not the intended recipient, please do not
read, copy, use or disclose this communication to others. Any
dissemination, distribution or copying of this communication other than to
the person or entity named above is strictly prohibited. If you have
received this communication in error, please immediately delete it from
your system.
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
From darnold at futurec.net Thu Jul 13 17:08:40 2006
From: darnold at futurec.net (David Arnold)
Date: Thu, 13 Jul 2006 17:08:40 -0700
Subject: [Live-devel] Streaming Raw PCM Audio Data
In-Reply-To: <444C5069.7000308@uwm.edu>
Message-ID:
Is there a subsession/fileSource class that can be used for unicast
streaming raw PCM audio? I need to stream audio from a device that
generates raw 16-bit PCM audio data. WAVAudioFileServerMediaSubsession
won't work.
Thank you,
Dave Arnold
Future Concepts, La Verne CA
The information contained in this electronic mail transmission is intended only for the use of the individual or entity named above and is privileged and confidential. If you are not the intended recipient, please do not read, copy, use or disclose this communication to others. Any dissemination, distribution or copying of this communication other than to the person or entity named above is strictly prohibited. If you have received this communication in error, please immediately delete it from your system.
From finlayson at live555.com Thu Jul 13 19:38:48 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 13 Jul 2006 19:38:48 -0700
Subject: [Live-devel] Playing streamed MP3 Audio with Quicktime
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060713193815.01f44050@live555.com>
At 01:20 PM 7/13/2006, you wrote:
>Dave,
>
>I'm new here, but I think that problem is mentioned on the livemedia
>website
>http://www.live555.com/liveMedia/faq.html#quicktime-player-mp3-bug
Yes. Everyone, *please* read the FAQ before posting a question.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jul 13 19:42:05 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 13 Jul 2006 19:42:05 -0700
Subject: [Live-devel] RTSP Documentation
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060713193858.01f3c008@live555.com>
At 08:58 AM 7/13/2006, you wrote:
>Hello, I've downloaded the Live555 library ("live555-latest.tar.gz") from
>http://www.live555.com/liveMedia/public/
>
>Right now I have a lot of questions about RTSP. I've been reading the RFC
>drafts from rtsp.org
>(http://www.rtsp.org/2003/drafts/draft05/draft-ietf-mmusic-rfc2326bis-05.txt),
> among other things.
>
>How can I find the version of the RFC that was used when writing my version
>of the library?
RFC 2326, which defined RTSP version 1.0. The newer "2326bis"
revision will be a new (& not quite backward compatible) RTSP
version, 2.0, that we (and most other people) don't (yet) support.
> How can I find out what parts of the RTSP standard are
>implemented?
The best way is to look at the code. If you have questions about
*specific* features of RTSP, you can ask on the mailing list. If you
have questions about 'trick mode' features, then please read the FAQ:
http://www.live555.com/liveMedia/faq.html
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jul 13 19:50:12 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 13 Jul 2006 19:50:12 -0700
Subject: [Live-devel] Streaming Raw PCM Audio Data
In-Reply-To:
References: <444C5069.7000308@uwm.edu>
Message-ID: <7.0.1.0.1.20060713194735.01f4e5a8@live555.com>
At 05:08 PM 7/13/2006, you wrote:
>Is there a subsession/fileSource class that can be used for unicast
>streaming raw PCM audio?
Yes, you can probably use "ByteStreamFileSource", provided that you
give it a proper "preferredFrameSize" parameter (should be a multiple
of 2 for mono audio; a multiple of 4 for stereo audio), and set the
"playTimePerFrame" parameter properly.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jul 13 19:51:04 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 13 Jul 2006 19:51:04 -0700
Subject: [Live-devel] TS Files and Sound
In-Reply-To: <194164850607131107j59819f26s9b9947108c5717f0@mail.gmail.co
m>
References: <194164850607050841k565b3734k709b98edcc8ad2e7@mail.gmail.com>
<194164850607060816n1c29ce08pf42b0639f33769d5@mail.gmail.com>
<7.0.1.0.1.20060706082130.02000930@live555.com>
<194164850607060924x5d275073g1e36ddcb4bc44806@mail.gmail.com>
<258726D5-AF56-42DE-A006-124FD40A1EBC@videolan.org>
<7.0.1.0.1.20060707212503.01ffec80@live555.com>
<194164850607101345w5b78e06u39a7fe5d84caa4b0@mail.gmail.com>
<7.0.1.0.1.20060711092055.01f6d378@live555.com>
<194164850607111347y78687fbfn8ecf7fa938ff4197@mail.gmail.com>
<7.0.1.0.1.20060711142502.01f93848@live555.com>
<194164850607131107j59819f26s9b9947108c5717f0@mail.gmail.com>
Message-ID: <7.0.1.0.1.20060713195036.01f44cf0@live555.com>
At 11:07 AM 7/13/2006, you wrote:
>Hi Ross,
>
>I tested the test.mpg that you provided, converted it using the
>testMPEG1or2ProgramToTransportStream program, and effectively the
>A110 STB played it correctly with sound.
>
>It is indeed strange why sound of other videos is not played at
>all. I tried with two other mpg program sources and none played
>sound. They were all converted using
>testMPEG1or2ProgramToTransportStream . You can find them here:
>
>http://208.131.130.36/coca.mpg ->
>http://208.131.130.36/coca.ts
>http://208.131.130.36/abue.mpg ->
>http://208.131.130.36/abue.ts
>
>Could it be some kind of encoding of sound with the streams I'm using?
I don't know; you'll have to ask Amino. It appears to be their bug.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From finlayson at live555.com Thu Jul 13 19:51:45 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 13 Jul 2006 19:51:45 -0700
Subject: [Live-devel] wis-streamer image freezing
In-Reply-To:
References:
Message-ID: <7.0.1.0.1.20060713195110.01f20f80@live555.com>
>Just got the latest live library (07-05 i believe) and the source
>for wis-streamer. Compiled the OSS WISGO driver. Used VLC to play
>the stream. Things work for about 30 seconds, then the image freezes
>and wis-streamer needs to be restarted
What audio and/or video codec(s) are you encoding/streaming?
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From andrew.voznytsa at gmail.com Fri Jul 14 06:40:34 2006
From: andrew.voznytsa at gmail.com (Andrew Voznytsa)
Date: Fri, 14 Jul 2006 16:40:34 +0300
Subject: [Live-devel] troubles with Vidiator
Message-ID: <44B79ED2.5010405@gmail.com>
Hello,
I built 3GPP client based on liveMedia library - liveMedia works
perfectly - pre-recorded and live content from Darwin and Helix servers
is received without any problems.
Recently I tried Vidiator and in case of live streaming (over 3G
channels) got problem - my MediaSink::afterGettingFrame() receives time
stamps (for audio frames) which are behind previous frame or +80 seconds
ahead or even more (rtpSource()->setPacketReorderingThresholdTime() was
up to 10 seconds). Problem exists in case if MPEG-4 video and MPEG-4
audio (LATM) is streamed and only in case of live streaming.
I tried capturing traffic with Ethereal and analyzing RTP timestamps -
they looks ok (monotony increasing). What I noticed that it depends on
RTP/RTCP receiving order - if video packets or audio RTCP comes first
then everything works ok.
I guess that it is somehow related to liveMedia internals. Could you
give me any idea where to dig? If you need I may provide Ethereal dumps
--
Best regards,
Andrew Voznytsa
From finlayson at live555.com Fri Jul 14 06:52:04 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 14 Jul 2006 06:52:04 -0700
Subject: [Live-devel] troubles with Vidiator
In-Reply-To: <44B79ED2.5010405@gmail.com>
References: <44B79ED2.5010405@gmail.com>
Message-ID: <7.0.1.0.1.20060714065031.01f2ff08@live555.com>
>What I noticed that it depends on
>RTP/RTCP receiving order - if video packets or audio RTCP comes first
>then everything works ok.
Note that any "presentationTime"s that are computed before the first
RTCP "SR" packet arrives will not necessarily be accurate. However,
once the first RTCP "SR" packet arrives, all subsequent
"presentationTime"s will be accurate (unless your server is buggy in some way).
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From ibaldine at rti.org Fri Jul 14 07:13:18 2006
From: ibaldine at rti.org (Baldine, Ilia)
Date: Fri, 14 Jul 2006 10:13:18 -0400
Subject: [Live-devel] wis-streamer image freezing
Message-ID:
Whatever is the default - I believe its MPEG4. Audio I'm not using, I
can try disabling it.
-i
-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, July 13, 2006 10:52 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] wis-streamer image freezing
>Just got the latest live library (07-05 i believe) and the source for
>wis-streamer. Compiled the OSS WISGO driver. Used VLC to play the
>stream. Things work for about 30 seconds, then the image freezes and
>wis-streamer needs to be restarted
What audio and/or video codec(s) are you encoding/streaming?
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
From ibaldine at rti.org Fri Jul 14 07:28:59 2006
From: ibaldine at rti.org (Baldine, Ilia)
Date: Fri, 14 Jul 2006 10:28:59 -0400
Subject: [Live-devel] wis-streamer image freezing
Message-ID:
Tried running 'wis-streamer -na' - same result. About 30 seconds of
streaming, then a freeze.
-ilia
-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, July 13, 2006 10:52 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] wis-streamer image freezing
>Just got the latest live library (07-05 i believe) and the source for
>wis-streamer. Compiled the OSS WISGO driver. Used VLC to play the
>stream. Things work for about 30 seconds, then the image freezes and
>wis-streamer needs to be restarted
What audio and/or video codec(s) are you encoding/streaming?
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
From andrew.voznytsa at gmail.com Fri Jul 14 07:43:56 2006
From: andrew.voznytsa at gmail.com (Andrew Voznytsa)
Date: Fri, 14 Jul 2006 17:43:56 +0300
Subject: [Live-devel] troubles with Vidiator
In-Reply-To: <7.0.1.0.1.20060714065031.01f2ff08@live555.com>
References: <44B79ED2.5010405@gmail.com>
<7.0.1.0.1.20060714065031.01f2ff08@live555.com>
Message-ID: <44B7ADAC.4050606@gmail.com>
Thank you Ross, it seems that this is reason of my problem.
If you have a minute - could you please give me hint how to detect when
the first RTCP "SR" packet was received?
BTW, I did small changes in liveMedia: in case if network cable was
removed from network card CPU usage raises to 100%. Inserting cable does
not lower CPU usage (It is related to Windows platform). I inserted
Sleep(1) somewhere and it helped. Maybe there is possible better fix but
if you'd I could send my patch as initial solution.
Best regards,
Andrew Voznytsa
Ross Finlayson wrote:
>> What I noticed that it depends on
>> RTP/RTCP receiving order - if video packets or audio RTCP comes first
>> then everything works ok.
>>
>
> Note that any "presentationTime"s that are computed before the first
> RTCP "SR" packet arrives will not necessarily be accurate. However,
> once the first RTCP "SR" packet arrives, all subsequent
> "presentationTime"s will be accurate (unless your server is buggy in some way).
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
From finlayson at live555.com Fri Jul 14 08:05:52 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 14 Jul 2006 08:05:52 -0700
Subject: [Live-devel] troubles with Vidiator
In-Reply-To: <44B7ADAC.4050606@gmail.com>
References: <44B79ED2.5010405@gmail.com>
<7.0.1.0.1.20060714065031.01f2ff08@live555.com>
<44B7ADAC.4050606@gmail.com>
Message-ID: <7.0.1.0.1.20060714080537.01f6a5f8@live555.com>
At 07:43 AM 7/14/2006, you wrote:
>Thank you Ross, it seems that this is reason of my problem.
>
>If you have a minute - could you please give me hint how to detect when
>the first RTCP "SR" packet was received?
You can use the function "RTPSource::hasBeenSynchronizedUsingRTCP()"
to test whether or not RTCP "Sender Report"s have started arriving.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From jiri.pinkava at vscht.cz Fri Jul 14 13:36:21 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Fri, 14 Jul 2006 22:36:21 +0200
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <7.0.1.0.1.20060706181637.01c90860@live555.com>
References: <44AD70AE.5090003@vscht.cz>
<7.0.1.0.1.20060706181637.01c90860@live555.com>
Message-ID: <20060714223621.ac204c3d.jiri.pinkava@vscht.cz>
On Thu, 06 Jul 2006 18:19:07 -0700
Ross Finlayson wrote:
> At 01:21 PM 7/6/2006, you wrote:
> >There are new release of my TS Framer/Simple Framer.
>
> Rather than simply sending code for this new class (which I have no
> plans to add to the library) without explanation, it might be more
> instructive if you explained what you think is wrong with the
> existing class, and what your new class does that supposedly improves upon it.
Sorry I have no experience with team development .....
>
> If there really is a problem with the existing code, then the best
> solution would be to fix that, because that's what everyone is going
> to get to see.
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
Problems and solutions for MPEG2TransportStreamFramer
I'm usign live555 to stream MPEG2 TS from DVB-T (digital video broadcast). Current MPEG2TransportStreamFramer implement both framer and timing, this is good for file sources, but not for live sources.
1. First goal is separate framing and timing, because for live source is necessary use other timer algorithm (current TS Framer/Timer cause packet lost and CPU overload for live source.)
2. I use TCP connection as source, this is true stream (is not framed), sometimes happen that half of frame is dropped by current framer. This is bad, part of frame must wait to be completed.
3. Current framing class does not handle (much pretty) errors inside stream (look only for sync byt in first TS packet and expect that all other is right).
4. next goal is control amount of TS packet present in frame by TS framing class (if I look at testMPEG2TransportStreamer this is now done by setting prefferedFrameSize in source, but this is little hacky). But this is minor issue.
There are most important reasons for major changes in MPEG2 TS framer, implemented in my new class.
From morgan.torvolt at gmail.com Sun Jul 16 06:50:59 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Sun, 16 Jul 2006 17:50:59 +0400
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <20060714223621.ac204c3d.jiri.pinkava@vscht.cz>
References: <44AD70AE.5090003@vscht.cz>
<7.0.1.0.1.20060706181637.01c90860@live555.com>
<20060714223621.ac204c3d.jiri.pinkava@vscht.cz>
Message-ID: <3cc3561f0607160650m18a7cc9bxda6c5a769746cbf3@mail.gmail.com>
I am not sure about this, but Is it not better to just drop the framer
of a live source? All the framer does is make sure the playout is done
at the correct time. If the source already has the correct timing, why
bother to re-calculate it?
-Morgan-
On 15/07/06, jiri.pinkava at vscht.cz wrote:
> On Thu, 06 Jul 2006 18:19:07 -0700
> Ross Finlayson wrote:
>
> > At 01:21 PM 7/6/2006, you wrote:
> > >There are new release of my TS Framer/Simple Framer.
> >
> > Rather than simply sending code for this new class (which I have no
> > plans to add to the library) without explanation, it might be more
> > instructive if you explained what you think is wrong with the
> > existing class, and what your new class does that supposedly improves upon it.
>
> Sorry I have no experience with team development .....
>
> >
> > If there really is a problem with the existing code, then the best
> > solution would be to fix that, because that's what everyone is going
> > to get to see.
> >
> >
> > Ross Finlayson
> > Live Networks, Inc. (LIVE555.COM)
> >
> >
> > _______________________________________________
> Problems and solutions for MPEG2TransportStreamFramer
> I'm usign live555 to stream MPEG2 TS from DVB-T (digital video broadcast). Current MPEG2TransportStreamFramer implement both framer and timing, this is good for file sources, but not for live sources.
>
> 1. First goal is separate framing and timing, because for live source is necessary use other timer algorithm (current TS Framer/Timer cause packet lost and CPU overload for live source.)
>
> 2. I use TCP connection as source, this is true stream (is not framed), sometimes happen that half of frame is dropped by current framer. This is bad, part of frame must wait to be completed.
>
> 3. Current framing class does not handle (much pretty) errors inside stream (look only for sync byt in first TS packet and expect that all other is right).
>
> 4. next goal is control amount of TS packet present in frame by TS framing class (if I look at testMPEG2TransportStreamer this is now done by setting prefferedFrameSize in source, but this is little hacky). But this is minor issue.
>
> There are most important reasons for major changes in MPEG2 TS framer, implemented in my new class.
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From barounis at ceid.upatras.gr Sun Jul 16 15:47:43 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Mon, 17 Jul 2006 01:47:43 +0300
Subject: [Live-devel] seg fault
In-Reply-To: <44B79ED2.5010405@gmail.com>
References: <44B79ED2.5010405@gmail.com>
Message-ID: <1153090063.44bac20fd565f@my.ceid.upatras.gr>
Hello to all members,
I have a very short question to ask:
Each time I am trying to add a member-data in the RTPSink class (header file)
although the compiler finds everything ok, I run the server and a seg fault
appears when the ~RTPSink (destructor) is called.
E.g I add the line "int a;" and the seg fault appears. Can you tell me what is
it going wrong ?
Thank u very much
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From finlayson at live555.com Sun Jul 16 18:02:58 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sun, 16 Jul 2006 18:02:58 -0700
Subject: [Live-devel] seg fault
In-Reply-To: <1153090063.44bac20fd565f@my.ceid.upatras.gr>
References: <44B79ED2.5010405@gmail.com>
<1153090063.44bac20fd565f@my.ceid.upatras.gr>
Message-ID: <7.0.1.0.1.20060716180145.01f25b00@live555.com>
At 03:47 PM 7/16/2006, you wrote:
>Hello to all members,
>
>I have a very short question to ask:
>
>Each time I am trying to add a member-data in the RTPSink class (header file)
>although the compiler finds everything ok, I run the server and a seg fault
>appears when the ~RTPSink (destructor) is called.
>
>E.g I add the line "int a;" and the seg fault appears. Can you tell
>me what is
>it going wrong ?
Most likely there's an old binary file somewhere that assumes the old
definition. I suggest doing a "make clean;make" everywhere.
Ross Finlayson
Live Networks, Inc. (LIVE555.COM)
From jiri.pinkava at vscht.cz Mon Jul 17 05:01:59 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Mon, 17 Jul 2006 14:01:59 +0200
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <194164850607061634r70714417r408424916a94b96e@mail.gmail.com>
References: <44AD70AE.5090003@vscht.cz>
<194164850607061410k64a690c2l9066d74abfa2f976@mail.gmail.com>
<194164850607061634r70714417r408424916a94b96e@mail.gmail.com>
Message-ID: <20060717140159.e934d464.jiri.pinkava@vscht.cz>
On Thu, 6 Jul 2006 16:34:16 -0700
"Napoleon Cornejo" wrote:
> Does this class add a PID for sound frames?
>
> I was having a problem with the Aminet 110 STB with sound. The streamed
> video played ok, but without sound, apparently because the sound had no PID.
>
> The way I was doing was using the MPEG2TransportStreamFromPESSource class.
>
This class have nothing to do with PIDs, this only do framing of MPEG2 TS.
There might be something wrong in MPEG2TSMultiplexor or around.
> Napoleon.
>
>
>
>
> > On 7/6/06, pinky wrote:
> >
> > > There are new release of my TS Framer/Simple Framer. They are tested and
> > working and broken TS stream, but more testing is willcomen and
> > necessary.
> >
> >
> > // A filter that passes through (unchanged) chunks that contain an
> > integral number
> > // of MPEG-2 Transport Stream packets, but returning (in
> > "fDurationInMicroseconds")
> > // an updated estimate of the time gap between chunks.
> > // C++ header
> >
> > #ifndef _MPEG2_TRANSPORT_STREAM_FRAMER_SIMPLE_HH
> > #define _MPEG2_TRANSPORT_STREAM_FRAMER_SIMPLE_HH
> >
> > #ifndef _FRAMED_FILTER_HH
> > #include "FramedFilter.hh"
> > #endif
> >
> > #define TRANSPORT_PACKET_SIZE 188
> >
> > class MPEG2TransportStreamFramerSimple: public FramedFilter {
> > public:
> >
>
> ........ MESSAGE LENGTH TRIMMED ......
>
From jiri.pinkava at vscht.cz Mon Jul 17 05:28:33 2006
From: jiri.pinkava at vscht.cz (jiri.pinkava at vscht.cz)
Date: Mon, 17 Jul 2006 14:28:33 +0200
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <3cc3561f0607160650m18a7cc9bxda6c5a769746cbf3@mail.gmail.com>
References: <44AD70AE.5090003@vscht.cz>
<7.0.1.0.1.20060706181637.01c90860@live555.com>
<20060714223621.ac204c3d.jiri.pinkava@vscht.cz>
<3cc3561f0607160650m18a7cc9bxda6c5a769746cbf3@mail.gmail.com>
Message-ID: <20060717142833.d9b3c839.jiri.pinkava@vscht.cz>
On Sun, 16 Jul 2006 17:50:59 +0400
"Morgan T?rvolt" wrote:
> I am not sure about this, but Is it not better to just drop the framer
> of a live source? All the framer does is make sure the playout is done
> at the correct time. If the source already has the correct timing, why
> bother to re-calculate it?
>
> -Morgan-
In many cases yes, it is. But from DVB-T (respective from incoming TCP stream) come data in "packs" of length cca 200-400kB and this (might) cause network overload and packet loss.
>
> On 15/07/06, jiri.pinkava at vscht.cz wrote:
> > On Thu, 06 Jul 2006 18:19:07 -0700
> > Ross Finlayson wrote:
> >
> > > At 01:21 PM 7/6/2006, you wrote:
> > > >There are new release of my TS Framer/Simple Framer.
> > >
> > > Rather than simply sending code for this new class (which I have no
> > > plans to add to the library) without explanation, it might be more
> > > instructive if you explained what you think is wrong with the
> > > existing class, and what your new class does that supposedly improves upon it.
> >
> > Sorry I have no experience with team development .....
> >
> > >
> > > If there really is a problem with the existing code, then the best
> > > solution would be to fix that, because that's what everyone is going
> > > to get to see.
> > >
> > >
> > > Ross Finlayson
> > > Live Networks, Inc. (LIVE555.COM)
> > >
> > >
> > > _______________________________________________
> > Problems and solutions for MPEG2TransportStreamFramer
> > I'm usign live555 to stream MPEG2 TS from DVB-T (digital video broadcast). Current MPEG2TransportStreamFramer implement both framer and timing, this is good for file sources, but not for live sources.
> >
> > 1. First goal is separate framing and timing, because for live source is necessary use other timer algorithm (current TS Framer/Timer cause packet lost and CPU overload for live source.)
> >
> > 2. I use TCP connection as source, this is true stream (is not framed), sometimes happen that half of frame is dropped by current framer. This is bad, part of frame must wait to be completed.
> >
> > 3. Current framing class does not handle (much pretty) errors inside stream (look only for sync byt in first TS packet and expect that all other is right).
> >
> > 4. next goal is control amount of TS packet present in frame by TS framing class (if I look at testMPEG2TransportStreamer this is now done by setting prefferedFrameSize in source, but this is little hacky). But this is minor issue.
> >
> > There are most important reasons for major changes in MPEG2 TS framer, implemented in my new class.
> > _______________________________________________
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> >
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
From barounis at ceid.upatras.gr Tue Jul 18 01:48:32 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Tue, 18 Jul 2006 11:48:32 +0300
Subject: [Live-devel] stopPlaying()
In-Reply-To: <7.0.1.0.1.20060524195000.01ec82d8@live555.com>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
Message-ID: <1153212512.44bca060a9fea@my.ceid.upatras.gr>
>
> To change the input source for a running stream, you should do the
> following, in order:
>
> sink->stopPlaying();
> Medium::close(oldSource);
> create newSource
> sink->startPlaying(newSource, ...);
Hello to all members,
before stopping the stream and before playing another stream I need to know the
exact frame in which the first video stopped. I also need a way to find the
respective frame in the new video file and then start playing. Can you tell me
please how I can implement this ?
Thank u very much
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From ymreddy at ssdi.sharp.co.in Tue Jul 18 04:10:55 2006
From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT))
Date: Tue, 18 Jul 2006 16:40:55 +0530
Subject: [Live-devel] SSRC Collision support in LIVE555
Message-ID: <7FB4685EA93D014C8E30AA087B66E7520267BB97@ssdimailsrvnt01.ssdi.sharp.co.in>
Hi Everybody,
Does LIVE555 support SSRC Collision and resolution. I see a comment "We
should really check for & handle >1 SSRCs being present #####" in RTCP.cpp
file.
Please provide the input.
Thanks and Regards
Y. Mallikharjuna Reddy
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From yossyd at nayos.com Tue Jul 18 07:04:39 2006
From: yossyd at nayos.com (Yossy Dreyfus)
Date: Tue, 18 Jul 2006 16:04:39 +0200
Subject: [Live-devel] DeviceSource
Message-ID: <000001c6aa73$1d14c820$570a1f0a@nayos.local>
I have an encode device that send me buffers with MPEG4 frames in real
time.
I need to send the buffers with RTSP.
How should I implement the DeviceSource class?
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From dicouw at gmail.com Tue Jul 18 06:52:32 2006
From: dicouw at gmail.com (dicou)
Date: Tue, 18 Jul 2006 14:52:32 +0100
Subject: [Live-devel] Rmvb streaming implementation
Message-ID: <2f4464440607180652s5028b8b1x32df30930bfda25@mail.gmail.com>
Hello everybody,
I just want to know if someone knows if it's possible to implement
the Rmvb streaming in this library,
i mean, at the present time, there are Mpeg1 Mpeg2 and Mpeg4, mp3,
is there a way to reuse the common function with rmvb and create/implement
the uncommon?
Thanks a lot.
--
Fred
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From bidibulle at operamail.com Tue Jul 18 08:16:34 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Tue, 18 Jul 2006 16:16:34 +0100
Subject: [Live-devel] RTCPInstance and RTP timestamp
Message-ID: <20060718151634.45EA77AE18@ws5-10.us4.outblaze.com>
Hi Ross,
I've struggled for a whisle on a synchronization problem between audio and video occuring with my streaming application (based on liveMedia library). I noticed that main players attach more importance to the timestamp information contained in the RTP-info header than to the timestamp information contained in the RTCP SRs. There is a case where the information sent in the RTP-info and the RTP timestamps of the very first timestamp of each track is wrong.
Let's say for example that the first packet sent for audio track is a RTP packet, the timestamp used in this packet will be the timestamp published in the RTP-info header, i.e. fTimestampBase (up to now, this is fine). However, if the first packet sent for the video track is a RTCP SR, this one will used as RTP timestamp the timestamp published in the RTP-info header, i.e. fTimestampBase. And then we have a problem with the next packet sent for video, which will be fTimestampBase + timestampIncrement.
The two tracks are now desynchronized (to be synchronized they should both use fTimestampBase as first timestamp).
I recognize my problem is quite rare (if you stream from a file, you can send RTP packets immediately, so RTCP packets will probably be sent after the first RTP packets) and also very difficult to resolve properly. IMO, the only acceptable way is to send RTCP packets only if a RTP packet has already been sent for the same track. This fix works for my particular problem and I would like to propose it as patch for the library. Here is the code :
In RTCPInstance::addReport(), replace :
if (fSink != NULL) {
addSR();
} else if (fSource != NULL) {
addRR();
}
with
if (fSink != NULL && fSink->haveComputedFirstTimestamp()) {
addSR();
} else if (fSource != NULL) {
addRR();
}
In RTPSink.hh, add public method to access private field fHaveComputedFirstTimestamp :
Boolean haveComputedFirstTimestamp() const { return fHaveComputedFirstTimestamp; }
Thank you in advance for your feedback
David
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 9 at http://www.opera.com
Powered by Outblaze
From fflood at eee.strath.ac.uk Tue Jul 18 09:43:02 2006
From: fflood at eee.strath.ac.uk (Frances Flood)
Date: Tue, 18 Jul 2006 17:43:02 +0100
Subject: [Live-devel] Playing streams on relay machine
Message-ID: <081913596FC02E4DB25A771862A592A91153A0@helios.eee.strath.ac.uk>
Hi,
I've got a program which receives audio and video streams and uses
live555 to relay them to other machines. However, I now want to be able
to play the streams (in sync if I've got both audio and video) on the
machine where my relay runs. Looking at a recent posting on "Streaming
from one source to multiple sink", I'm wondering if that's what I need
to be doing? Could you also point me to a suitable example in the test
programs for the sink I need to play the streams in, say, VLC? I've
tried various ones, but just seem to get port errors, so don't know
whether I've got the wrong example or am doing something else wrong.
Many thanks,
Frances
From andrew.voznytsa at gmail.com Wed Jul 19 03:57:56 2006
From: andrew.voznytsa at gmail.com (Andrew Voznytsa)
Date: Wed, 19 Jul 2006 13:57:56 +0300
Subject: [Live-devel] troubles with Vidiator
In-Reply-To: <7.0.1.0.1.20060714080537.01f6a5f8@live555.com>
References: <44B79ED2.5010405@gmail.com> <7.0.1.0.1.20060714065031.01f2ff08@live555.com> <44B7ADAC.4050606@gmail.com>
<7.0.1.0.1.20060714080537.01f6a5f8@live555.com>
Message-ID: <44BE1034.9080801@gmail.com>
Ross Finlayson wrote:
> At 07:43 AM 7/14/2006, you wrote:
>
>> Thank you Ross, it seems that this is reason of my problem.
>>
>> If you have a minute - could you please give me hint how to detect when
>> the first RTCP "SR" packet was received?
>>
>
> You can use the function "RTPSource::hasBeenSynchronizedUsingRTCP()"
> to test whether or not RTCP "Sender Report"s have started arriving.
>
Unfortunately I'm unable to do this in case of AMR audio (or I'm missing
something).
If I'm receiving AMR then AMRDeinterleaver is used as source of AMR
audio. AMRDeinterleaver::isRTPSource() returns False which means that
I'm unable to call RTPSource::hasBeenSynchronizedUsingRTCP() in this
case. Actually AMRDeinterleaver receives data using RawAMRRTPSource
(through FramedSource* fInputSource class member). So IMO
AMRDeinterleaver should have some method to return fInputSource or
implement RTPSource and forward all calls to fInputSource. I don't see
how to get access to AMR's RTPSource::hasBeenSynchronizedUsingRTCP()
otherwise..
From finlayson at live555.com Wed Jul 19 05:07:55 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 19 Jul 2006 05:07:55 -0700
Subject: [Live-devel] RTCPInstance and RTP timestamp
In-Reply-To: <20060718151634.45EA77AE18@ws5-10.us4.outblaze.com>
References: <20060718151634.45EA77AE18@ws5-10.us4.outblaze.com>
Message-ID:
>I recognize my problem is quite rare (if you stream from a file, you
>can send RTP packets immediately, so RTCP packets will probably be
>sent after the first RTP packets) and also very difficult to resolve
>properly. IMO, the only acceptable way is to send RTCP packets only
>if a RTP packet has already been sent for the same track. This fix
>works for my particular problem and I would like to propose it as
>patch for the library.
That looks reasonable. It will be included in the next release of
the software.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From kushal.dalal at einfochips.com Wed Jul 19 06:27:54 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Wed, 19 Jul 2006 18:57:54 +0530
Subject: [Live-devel] H.264 support
Message-ID: <200607191328.k6JDS3ga090468@ns.live555.com>
Hi All,
It seems from latest source code and changelog, live supports H.264
streaming over RTP.
I don't find any test application for that.
Even I don't find session/subsession classes for H.264.
Does live supports H.264 streaming?
If yes then how can I use it?
Thanks
Kushal
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From finlayson at live555.com Wed Jul 19 10:46:41 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 19 Jul 2006 10:46:41 -0700
Subject: [Live-devel] H.264 support
In-Reply-To: <200607191328.k6JDS3ga090468@ns.live555.com>
References: <200607191328.k6JDS3ga090468@ns.live555.com>
Message-ID:
>Hi All,
>
>It seems from latest source code and changelog, live supports H.264
>streaming over RTP.
>I don't find any test application for that.
That's because the library doesn't yet support reading H.264 (or
anything else) from MPEG-4 format files.
However, if you have a source class that delivers discrete H.264 NAL
units, then you can use the library to stream H.264 video over RTP.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From finlayson at live555.com Wed Jul 19 11:09:19 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 19 Jul 2006 11:09:19 -0700
Subject: [Live-devel] SSRC Collision support in LIVE555
In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520267BB97@ssdimailsrvnt01.ssdi.sharp.co.in
>
References: <7FB4685EA93D014C8E30AA087B66E7520267BB97@ssdimailsrvnt01.ssdi.sharp.co.in
>
Message-ID:
>Hi Everybody,
>
>Does LIVE555 support SSRC Collision and resolution.
No, not at present. It has not yet been implemented in the libraries.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Wed Jul 19 17:31:54 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 19 Jul 2006 17:31:54 -0700
Subject: [Live-devel] troubles with Vidiator
In-Reply-To: <44BE1034.9080801@gmail.com>
References: <44B79ED2.5010405@gmail.com>
<7.0.1.0.1.20060714065031.01f2ff08@live555.com>
<44B7ADAC.4050606@gmail.com>
<7.0.1.0.1.20060714080537.01f6a5f8@live555.com>
<44BE1034.9080801@gmail.com>
Message-ID:
> > You can use the function "RTPSource::hasBeenSynchronizedUsingRTCP()"
>> to test whether or not RTCP "Sender Report"s have started arriving.
>>
>Unfortunately I'm unable to do this in case of AMR audio (or I'm missing
>something).
>
>If I'm receiving AMR then AMRDeinterleaver is used as source of AMR
>audio. AMRDeinterleaver::isRTPSource() returns False which means that
>I'm unable to call RTPSource::hasBeenSynchronizedUsingRTCP() in this
>case. Actually AMRDeinterleaver receives data using RawAMRRTPSource
>(through FramedSource* fInputSource class member). So IMO
>AMRDeinterleaver should have some method to return fInputSource or
>implement RTPSource and forward all calls to fInputSource
No problem. If you look at the call to
"AMRAudioRTPSource::createNew()", you'll see that there's a result
parameter "resultRTPSource" which returns the "RTPSource" object
behind this.
Also, if you are accessing the stream using RTSP, you will have
created a "MediaSubsession" object for this stream. You can then
call its "rtpSource()" member function to get the "RTPSource" object.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From kushal.dalal at einfochips.com Wed Jul 19 23:28:00 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Thu, 20 Jul 2006 11:58:00 +0530
Subject: [Live-devel] H.264 support
In-Reply-To:
Message-ID: <200607200628.k6K6S9UR064692@ns.live555.com>
Ross,
Any plan to provide that in near future?
I don't know much about codec functionalities.
So I have few more queries.
Does H.264 codecs provide discrete NAL units or they need to be abstracted
from encoded frame?
I am currently getting encoded frame from my codec when I give raw video
data as input.
Thanks
Kushal
_____
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Ross Finlayson
Sent: Wednesday, July 19, 2006 11:17 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] H.264 support
Hi All,
It seems from latest source code and changelog, live supports H.264
streaming over RTP.
I don't find any test application for that.
That's because the library doesn't yet support reading H.264 (or anything
else) from MPEG-4 format files.
However, if you have a source class that delivers discrete H.264 NAL units,
then you can use the library to stream H.264 video over RTP.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From Heiko.Simon at iai.fzk.de Thu Jul 20 00:59:29 2006
From: Heiko.Simon at iai.fzk.de (Simon, Heiko)
Date: Thu, 20 Jul 2006 09:59:29 +0200
Subject: [Live-devel] Streaming from a live Source
Message-ID: <008F01C663C265459DE305F80366A4CE1B538C@iai-exchange.iai.fzk.de>
Hi there!
I would like to stream live video via LIVE555.
I want to use ffmpeg-libraries to encode it.
So far I only encode a test-image over and over again and feed it to LIVE555,
as described in: http://www.live555.com/liveMedia/faq.html#liveInput
I used the option using:
"For a model of how to do that, see "liveMedia/DeviceSource.cpp" (and "liveMedia/include/DeviceSource.hh"). You will need to fill in parts of this code to do the actual read from your encoder."
So far I encode the single pictures and hand the resulting frames to LIVE within the deliverframe()- method.
The programm compiles, starts streaming and so on but I can?t use receive anything with vlc as a client. VLC works as a client for the other examples of LIVE, where videofiles are streamed. So I thought I could just change on of the server- examples, so it streams my testimages instead of the file. I?m a bit confused therefore and I realized that not everything is clear to me:
One thing I?m not sure about is whether "framed source" means video- frames, or frames of data or something.
I take it, it?s video-frames. Is that right?
If so, here?s another one:
Does LIVE handle the packetizing of the frames? I actually assumed that LIVE would want to have Videostreams within a streamable packetformat like mp4 or so..
Instead it seams that I can just supply it with pure encoded frames. Is this ok like that?
If so, how can a client tell whether a received frame is encoded for example with divx, h263, or any other codec?
Thanks so far.
Feel free to ask me if I haven't been clear enough.
CU
Heiko
From arcon-media at gmx.net Thu Jul 20 02:31:07 2006
From: arcon-media at gmx.net (arcon media)
Date: Thu, 20 Jul 2006 11:31:07 +0200
Subject: [Live-devel] Livestream from IP-Cam to HelixServer???
Message-ID:
Hello,
Can I make a 3gp Stream from an IP-Camera. The Source from the IP-Camera is
a MPEG4-Stream.
I must receive the mpeg4 stream from Camera - make a 3gp with width and
height - send to a Helix Server.
Is this possible with the C++ libraries for multimedia streaming from
Live555? Which libraries do I need?
Can you help me?
Thank u
Brian
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From stas at mer.co.il Thu Jul 20 09:18:23 2006
From: stas at mer.co.il (Stas Desyatnlkov)
Date: Thu, 20 Jul 2006 18:18:23 +0200
Subject: [Live-devel] Recursion
Message-ID: <8F5A28107CB68247BB8E3D3EE626F54F01D41C36@fs1.mertree.mer.co.il>
Hi All,
In my sink class I have the following code:
G729RtpSink::afterGettingFrame1(void* clientData, unsigned frameSize,
unsigned truncBytes, timeval& pts,
unsigned duration_ms)
{
char stuff[1024];
...
...
continuePlaying();
}
I get stack overflow error if I allocate things on stack in
afterGettingFrame1.
It usually happens when a lot of data arrives on my sink/source port.
Does anyone experienced this behavior ? Any ideas how to solve it ?
Regards,
Stas
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From barounis at ceid.upatras.gr Thu Jul 20 12:02:24 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Thu, 20 Jul 2006 22:02:24 +0300
Subject: [Live-devel] switching files
In-Reply-To: <7.0.1.0.1.20060716180145.01f25b00@live555.com>
References: <44B79ED2.5010405@gmail.com>
<1153090063.44bac20fd565f@my.ceid.upatras.gr>
<7.0.1.0.1.20060716180145.01f25b00@live555.com>
Message-ID: <1153422144.44bfd340120fb@my.ceid.upatras.gr>
Hello to everybody on the list,
While giving the RTSPServer the ability to switch files during the same session
there must be found a way to keep track of the frames that have already been
sent (from the first file) so as to start sending the next-to-be-played frame
from the new file.
Ross can u please give me any suggestions.
Thank u very much
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From yunjnz at yahoo.com Thu Jul 20 19:59:53 2006
From: yunjnz at yahoo.com (yj)
Date: Thu, 20 Jul 2006 19:59:53 -0700 (PDT)
Subject: [Live-devel] MPEG4 ES Video Trick mode support
In-Reply-To: <44AD70AE.5090003@vscht.cz>
Message-ID: <20060721025953.38811.qmail@web35802.mail.mud.yahoo.com>
Hi,
I want to implement MPEG4 ES video Trick mode in the
Live555 Library,
are there any existing solution for this feature?
else how to do?
Also did Live555 Media plan to support Trick mode in
the future?
thank you in advance.
sean.
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
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From arcon-media at gmx.net Thu Jul 20 23:38:36 2006
From: arcon-media at gmx.net (arcon media)
Date: Fri, 21 Jul 2006 08:38:36 +0200
Subject: [Live-devel] RTP to RTCP for Helix
Message-ID:
Hello,
I use the VLC of videolan.org as Encoder for Live streams. These streams are
then sent as RTP. These Live streams shall be sent to a helix server.
Unfortunately, the VLC cannot make any RTCP packages. These are needed by
the helix server.
The Helixcommuntity wrote me:
"You could write a program that receives and relays VLC's RTP packets to the
server and generates appropriate RTCP packets and sends those, too. "
Is this possible with live libraries?
Can I make one kind of transducer with the help of live libraries after the
VLC? How can I solve the task with the help of live libraries?
Thank you for your help.
Brian
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From finlayson at live555.com Fri Jul 21 00:40:13 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 21 Jul 2006 00:40:13 -0700
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
References:
Message-ID:
>Hello,
>I use the VLC of videolan.org as Encoder for Live streams. These
>streams are then sent as RTP. These Live streams shall be sent to a
>helix server. Unfortunately, the VLC cannot make any RTCP packages.
That's because VLC - when used as a RTP *transmitter* - uses a
quick-and-dirty, custom implementation of RTP (*not* the "LIVE555
Streaming Media" code) that does not include RTCP. (VLC uses the
LIVE555 code only when it is used as a receiver/player.)
>These are needed by the helix server.
>The Helixcommuntity wrote me:
>"You could write a program that receives and relays VLC's RTP
>packets to the server and generates appropriate RTCP packets and
>sends those, too. "
>Is this possible with live libraries?
Perhaps. But it would be much better (and probably simpler) to use
our own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an
encoder.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From tjc103 at ecs.soton.ac.uk Fri Jul 21 01:14:20 2006
From: tjc103 at ecs.soton.ac.uk (tjc103 at ecs.soton.ac.uk)
Date: Fri, 21 Jul 2006 09:14:20 +0100
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
References:
Message-ID: <1153469660.44c08cdc8e2e0@webmail.soton.ac.uk>
Quoting Ross Finlayson :
> >Hello,
> >I use the VLC of videolan.org as Encoder for Live streams. These
> >streams are then sent as RTP. These Live streams shall be sent to a
> >helix server. Unfortunately, the VLC cannot make any RTCP packages.
>
> That's because VLC - when used as a RTP *transmitter* - uses a
> quick-and-dirty, custom implementation of RTP (*not* the "LIVE555
> Streaming Media" code) that does not include RTCP. (VLC uses the
> LIVE555 code only when it is used as a receiver/player.)
Just out of curiosity, does VLS suffer from this problem as well.
> >These are needed by the helix server.
> >The Helixcommuntity wrote me:
> >"You could write a program that receives and relays VLC's RTP
> >packets to the server and generates appropriate RTCP packets and
> >sends those, too. "
> >Is this possible with live libraries?
>
> Perhaps. But it would be much better (and probably simpler) to use
> our own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an
> encoder.
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
From arcon-media at gmx.net Fri Jul 21 02:18:30 2006
From: arcon-media at gmx.net (arcon media)
Date: Fri, 21 Jul 2006 11:18:30 +0200
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
Message-ID:
> Perhaps. But it would be much better (and probably simpler) to use our
own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an encoder.
Thanks for your fast answer. It's better with a own server implementation,
but our company has bought a licence for Helix. This is my problem. I am
looking for a solution for my problem.
Brian
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From kushal.dalal at einfochips.com Fri Jul 21 02:21:34 2006
From: kushal.dalal at einfochips.com (Kushal Dalal)
Date: Fri, 21 Jul 2006 14:51:34 +0530
Subject: [Live-devel] Streaming from a live Source
In-Reply-To: <008F01C663C265459DE305F80366A4CE1B538C@iai-exchange.iai.fzk.de>
Message-ID: <200607210921.k6L9LgQW035048@ns.live555.com>
Have you implemented session class "LiveVideoServerMediaSubsession" like
"FileServerMediaSubsession"?
What modification you did with "testOnDamandRTSPServer.cpp"?
- Kushal
-----Original Message-----
From: live-devel-bounces at ns.live555.com
[mailto:live-devel-bounces at ns.live555.com] On Behalf Of Simon, Heiko
Sent: Thursday, July 20, 2006 1:29 PM
To: live-devel at ns.live555.com
Subject: [Live-devel] Streaming from a live Source
Hi there!
I would like to stream live video via LIVE555.
I want to use ffmpeg-libraries to encode it.
So far I only encode a test-image over and over again and feed it to
LIVE555,
as described in: http://www.live555.com/liveMedia/faq.html#liveInput
I used the option using:
"For a model of how to do that, see "liveMedia/DeviceSource.cpp" (and
"liveMedia/include/DeviceSource.hh"). You will need to fill in parts of this
code to do the actual read from your encoder."
So far I encode the single pictures and hand the resulting frames to LIVE
within the deliverframe()- method.
The programm compiles, starts streaming and so on but I can?t use receive
anything with vlc as a client. VLC works as a client for the other examples
of LIVE, where videofiles are streamed. So I thought I could just change on
of the server- examples, so it streams my testimages instead of the file.
I?m a bit confused therefore and I realized that not everything is clear to
me:
One thing I?m not sure about is whether "framed source" means video- frames,
or frames of data or something.
I take it, it?s video-frames. Is that right?
If so, here?s another one:
Does LIVE handle the packetizing of the frames? I actually assumed that LIVE
would want to have Videostreams within a streamable packetformat like mp4 or
so..
Instead it seams that I can just supply it with pure encoded frames. Is this
ok like that?
If so, how can a client tell whether a received frame is encoded for example
with divx, h263, or any other codec?
Thanks so far.
Feel free to ask me if I haven't been clear enough.
CU
Heiko
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
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From glen at lincor.com Fri Jul 21 02:33:30 2006
From: glen at lincor.com (Glen Gray)
Date: Fri, 21 Jul 2006 10:33:30 +0100
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
References:
Message-ID: <44C09F6A.9010305@lincor.com>
Hey Brian,
arcon media wrote:
>> Perhaps. But it would be much better (and probably simpler) to use
> our own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an
> encoder.
>
>
>
> Thanks for your fast answer. It?s better with a own server
> implementation, but our company has bought a licence for Helix. This is
> my problem. I am looking for a solution for my problem.
>
I think Ross meant that you should use the Live555 RTSP/RTP server in
conjunction with FFMPEG to get the correct live streams TO the Helix
server. i.e. As opposed to using VLC to do the streaming to the server.
--
Glen Gray Digital Depot, Thomas Street
Senior Software Engineer Dublin 8, Ireland
Lincor Solutions Ltd. Ph: +353 (0) 1 4893682
From ymreddy at ssdi.sharp.co.in Fri Jul 21 03:45:56 2006
From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT))
Date: Fri, 21 Jul 2006 16:15:56 +0530
Subject: [Live-devel] Getting source IP address
Message-ID: <7FB4685EA93D014C8E30AA087B66E7520267BB9E@ssdimailsrvnt01.ssdi.sharp.co.in>
Hi Everybody,
I am trying to stream MPEG2 data from server to client in a unicast
environment. In the server side file, testMPEG1or2VideoStreamer.cpp, I have
given the client IP address. In the client side file,
testMPEG1or2VideoReceiver.cpp, again, I have given the client side IP
address.
When I receive the packets in the client side, I would like to get the
source IP address and port number from where the packets are coming. I have
added the following lines in networkReadHandler() function
MultiFramedRTPSource.cpp file.
******************************************************************
const struct in_addr in_addr1= source->RTPgs()->groupAddress();
char *ip = inet_ntoa(in_addr1);
const portNumBits port= source->RTPgs()->portnum();
*******************************************************************
In the above statement, I am getting the IP address (variable ip) of the
client machine instead of server. Port number is also same.
How do I get the server IP address. I am doing this for SSRC collision
detection and resoultion as part of my project work.
Thanks and Regards
Y. Mallikharjuna Reddy
From morgan.torvolt at gmail.com Fri Jul 21 04:44:46 2006
From: morgan.torvolt at gmail.com (=?ISO-8859-1?Q?Morgan_T=F8rvolt?=)
Date: Fri, 21 Jul 2006 15:44:46 +0400
Subject: [Live-devel] MPEG2TransportStreamFramer enhancement
In-Reply-To: <20060717142833.d9b3c839.jiri.pinkava@vscht.cz>
References: <44AD70AE.5090003@vscht.cz>
<7.0.1.0.1.20060706181637.01c90860@live555.com>
<20060714223621.ac204c3d.jiri.pinkava@vscht.cz>
<3cc3561f0607160650m18a7cc9bxda6c5a769746cbf3@mail.gmail.com>
<20060717142833.d9b3c839.jiri.pinkava@vscht.cz>
Message-ID: <3cc3561f0607210444n17c918a0k35e943a796eab287@mail.gmail.com>
I see. I would buffer the data then, and use decoding timestamp to
calculate the datarate.
-Morgan-
On 17/07/06, jiri.pinkava at vscht.cz wrote:
> On Sun, 16 Jul 2006 17:50:59 +0400
> "Morgan T?rvolt" wrote:
>
> > I am not sure about this, but Is it not better to just drop the framer
> > of a live source? All the framer does is make sure the playout is done
> > at the correct time. If the source already has the correct timing, why
> > bother to re-calculate it?
> >
> > -Morgan-
>
> In many cases yes, it is. But from DVB-T (respective from incoming TCP stream) come data in "packs" of length cca 200-400kB and this (might) cause network overload and packet loss.
>
> >
> > On 15/07/06, jiri.pinkava at vscht.cz wrote:
> > > On Thu, 06 Jul 2006 18:19:07 -0700
> > > Ross Finlayson wrote:
> > >
> > > > At 01:21 PM 7/6/2006, you wrote:
> > > > >There are new release of my TS Framer/Simple Framer.
> > > >
> > > > Rather than simply sending code for this new class (which I have no
> > > > plans to add to the library) without explanation, it might be more
> > > > instructive if you explained what you think is wrong with the
> > > > existing class, and what your new class does that supposedly improves upon it.
> > >
> > > Sorry I have no experience with team development .....
> > >
> > > >
> > > > If there really is a problem with the existing code, then the best
> > > > solution would be to fix that, because that's what everyone is going
> > > > to get to see.
> > > >
> > > >
> > > > Ross Finlayson
> > > > Live Networks, Inc. (LIVE555.COM)
> > > >
> > > >
> > > > _______________________________________________
> > > Problems and solutions for MPEG2TransportStreamFramer
> > > I'm usign live555 to stream MPEG2 TS from DVB-T (digital video broadcast). Current MPEG2TransportStreamFramer implement both framer and timing, this is good for file sources, but not for live sources.
> > >
> > > 1. First goal is separate framing and timing, because for live source is necessary use other timer algorithm (current TS Framer/Timer cause packet lost and CPU overload for live source.)
> > >
> > > 2. I use TCP connection as source, this is true stream (is not framed), sometimes happen that half of frame is dropped by current framer. This is bad, part of frame must wait to be completed.
> > >
> > > 3. Current framing class does not handle (much pretty) errors inside stream (look only for sync byt in first TS packet and expect that all other is right).
> > >
> > > 4. next goal is control amount of TS packet present in frame by TS framing class (if I look at testMPEG2TransportStreamer this is now done by setting prefferedFrameSize in source, but this is little hacky). But this is minor issue.
> > >
> > > There are most important reasons for major changes in MPEG2 TS framer, implemented in my new class.
> > > _______________________________________________
> > > live-devel mailing list
> > > live-devel at lists.live555.com
> > > http://lists.live555.com/mailman/listinfo/live-devel
> > >
> > _______________________________________________
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From arcon-media at gmx.net Fri Jul 21 04:44:31 2006
From: arcon-media at gmx.net (arcon media)
Date: Fri, 21 Jul 2006 13:44:31 +0200
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To: <44C09F6A.9010305@lincor.com>
Message-ID:
Hello,
How can I do this? Can you give me a solution trial?
Thanks
Brian
From finlayson at live555.com Fri Jul 21 04:56:47 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 21 Jul 2006 04:56:47 -0700
Subject: [Live-devel] Getting source IP address
In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520267BB9E@ssdimailsrvnt01.ssdi.sharp.co.in
>
References: <7FB4685EA93D014C8E30AA087B66E7520267BB9E@ssdimailsrvnt01.ssdi.sharp.co.in
>
Message-ID:
>When I receive the packets in the client side, I would like to get the
>source IP address and port number from where the packets are coming. I have
>added the following lines in networkReadHandler() function
>MultiFramedRTPSource.cpp file.
>
>******************************************************************
>
> const struct in_addr in_addr1= source->RTPgs()->groupAddress();
> char *ip = inet_ntoa(in_addr1);
>
> const portNumBits port= source->RTPgs()->portnum();
>
>*******************************************************************
>
>In the above statement, I am getting the IP address (variable ip) of the
>client machine instead of server. Port number is also same.
>
>How do I get the server IP address.
You do this my modifying "MultiFramedRTPSource.cpp", but not in the
way that you have done.
Instead, note the "fromAddress" parameter to the call to
"handleRead()" in "BufferedPacket::fillInData()". This (result)
parameter will contain the source IP address and port for the
received RTP packet.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From andrew.voznytsa at gmail.com Fri Jul 21 07:59:23 2006
From: andrew.voznytsa at gmail.com (Andrew Voznytsa)
Date: Fri, 21 Jul 2006 17:59:23 +0300
Subject: [Live-devel] troubles with Vidiator
In-Reply-To:
References: <44B79ED2.5010405@gmail.com> <7.0.1.0.1.20060714065031.01f2ff08@live555.com> <44B7ADAC.4050606@gmail.com> <7.0.1.0.1.20060714080537.01f6a5f8@live555.com> <44BE1034.9080801@gmail.com>
Message-ID: <44C0EBCB.5000103@gmail.com>
Ross Finlayson wrote:
> Also, if you are accessing the stream using RTSP, you will have
> created a "MediaSubsession" object for this stream. You can then
> call its "rtpSource()" member function to get the "RTPSource" object.
>
Simply an overlook on my part :( I'm sorry for bothering.
From tjc103 at ecs.soton.ac.uk Fri Jul 21 08:05:50 2006
From: tjc103 at ecs.soton.ac.uk (tjc103 at ecs.soton.ac.uk)
Date: Fri, 21 Jul 2006 16:05:50 +0100
Subject: [Live-devel] Running openRTSP
Message-ID: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
Hi,
I'm trying to integrate the openRTSP program into an existing application I
have. Initially while compiling I had a problem with NetCommon.h on the
line :
#define SOCKLEN_T int
changing this to :
#define SOCKLEN_T unsigned int
solved this completely. However the crux of my problem is with the PThreads.
At the moment I call the main (renamed to ip_main) function of playCommon
with pthread_create. Then when i want to end the openRTSP thread I issue a
pthread_kill(thread, SIGHUP) and wait for the thread to join. I've altered
the shutdown routine in playCommon to do a pthread_exit rather than exit.
This appears to work fine for starting and stopping the thread once, however
if I have qosMeasurementIntervalMS set to anything other than 0 I get a
segfault trying to start the thread again (it segfaults in the function
RTPSource::receptionStatsDB). I thought that this maybe due to some
variable not being cleared properly on shutdown of the openRTSP thread, but
env->taskScheduler().unscheduleDelayedTask(qosMeasurementTimerTask) is being
called fine.
If I do have qosMeasurementIntervalMS set to 0, then I can start and stop
threads fine, but there is an intermitent segfault while executing the
command pthread_exit. Doing a backtrace on this points to
pthread_mutex_unlock.
I'm not using any mutex locks at all (AFAIK I don't need to). However, just
to be sure I implemented pthread_mutex_lock at the start of openRTSP and
pthread_mutex_unlock just before the pthread_exit command is called - It
never segfaults on the pthread_mutex_unlock command I added, but will still
occasionly segfault on the pthread_exit command just after.
Does anybody have any experience with this? Is there a better way to start
and stop openRTSP (or playCommon) than using signals like hangup? I'm
guessing it may not be nescessary to stop it all, and have it idle in
someway.
Thanks
Theo
From dsmurl at yahoo.com Fri Jul 21 11:27:32 2006
From: dsmurl at yahoo.com (Sam b)
Date: Fri, 21 Jul 2006 11:27:32 -0700 (PDT)
Subject: [Live-devel] Compiling darwin injector example
Message-ID: <20060721182732.98943.qmail@web38401.mail.mud.yahoo.com>
I am trying to compile the test programs. I can get
all the objects in "livemedia" to compile in visual
studio 6.0 but when I try to compile the programs
under "testProgs"
by importing the testProgs.mak file into VS 6.0, it
gives me the following errors.
NMAKE : fatal error U1073: don't know how to make
'../groupsock/libgroupsock.lib'
Stop.
Error executing NMAKE.
testProgs1.exe - 1 error(s), 0 warning(s)
How can i get this to compile? Also, does anyone have
any suggestions on how to modify it to except an h264
stream? Any help would be great. Thanks
- Sam
__________________________________________________
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From ibaldine at rti.org Fri Jul 21 11:45:04 2006
From: ibaldine at rti.org (Baldine, Ilia)
Date: Fri, 21 Jul 2006 14:45:04 -0400
Subject: [Live-devel] Patch for wis-streamer
Message-ID:
This is a small patch to wis-streamer app to make it work on a big-endian ARM (using snapgear 3.3.0, linux-2.6.x). It declares videoType explicitly to be unsigned long long (64-bit long) instead of 32-bit int. Internally in the wis-go7007 driver, during ioctl call, it is compared to a 64-bit u_int, and for some reason the arm compiler does not explicitly upconvert from 32-bit declared int to 64-bit final value, so the ioctl returns EINVAL and breaks the program, unable to set video type. With the patch all works well.
I suppose it is equally possible to patch the driver, but patching user-space seems easier.
-ilia
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From finlayson at live555.com Fri Jul 21 13:35:46 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 21 Jul 2006 13:35:46 -0700
Subject: [Live-devel] Patch for wis-streamer
In-Reply-To:
References:
Message-ID:
>This is a small patch to wis-streamer app to make it work on a
>big-endian ARM (using snapgear 3.3.0, linux-2.6.x). It declares
>videoType explicitly to be unsigned long long (64-bit long) instead
>of 32-bit int. Internally in the wis-go7007 driver, during ioctl
>call, it is compared to a 64-bit u_int, and for some reason the arm
>compiler does not explicitly upconvert from 32-bit declared int to
>64-bit final value, so the ioctl returns EINVAL and breaks the
>program, unable to set video type. With the patch all works well.
Thanks. I will add this to the released code shortly.
>I suppose it is equally possible to patch the driver
Actually, patching the driver would be better, because it would
prevent other applications from running into the same problem. If
you can generate an equivalent patch for the driver code, then please
send it to me, and I'll see that it gets added to WIS (now called
Micronas)'s released driver code. (I'm in close contact with these
folks.)
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Fri Jul 21 13:52:49 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 21 Jul 2006 13:52:49 -0700
Subject: [Live-devel] Running openRTSP
In-Reply-To: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
Message-ID:
>This appears to work fine for starting and stopping the thread once, however
>if I have qosMeasurementIntervalMS set to anything other than 0 I get a
>segfault trying to start the thread again (it segfaults in the function
>RTPSource::receptionStatsDB). I thought that this maybe due to some
>variable not being cleared properly on shutdown of the openRTSP thread
Yes - that's probably the case. Note that the "openRTSP" code was
intended to be run as a single application - not run more than once
in the same process.
I think your use of threads is a red herring here (I assume, BTW,
that you have read the FAQ entries that discuss threads). I suspect
that you'd run into similar problems (but much easier to debug) if
you were to call the openRTSP code more than once - simply as a
procedure. So that's what I suggest you do, for now.
It's possible that there is some more cleanup code that should be
added to "shutdown()"; if you find it, please let us know.
Remember, You Have Complete Source Code.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From sdhays.neon.com.tw at gmail.com Fri Jul 21 19:37:24 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Sat, 22 Jul 2006 10:37:24 +0800
Subject: [Live-devel] Compiling darwin injector example
In-Reply-To: <20060721182732.98943.qmail@web38401.mail.mud.yahoo.com>
References: <20060721182732.98943.qmail@web38401.mail.mud.yahoo.com>
Message-ID: <9866ce4f0607211937l67cd9bedof10fb518012d8697@mail.gmail.com>
Have you compiled '../groupsock/libgroupsock.lib'?
Scott
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From spacelis at gmail.com Fri Jul 21 22:05:45 2006
From: spacelis at gmail.com (SpaceLi)
Date: Sat, 22 Jul 2006 13:05:45 +0800
Subject: [Live-devel] Can I run LIVE.COM Streaming Media on my board of Nios
with uClinux?
Message-ID: <4f8ff5c10607212205j32324463x250a5b4ad4beedbf@mail.gmail.com>
I have searched the miling list on live555.com, but there is no hint
for nios2. How can I port it to nios2. And I don't know wether the it
is the same with ARM.
Here is bin of the compiler direcories
nios2-linux-addr2line nios2-linux-readelf nios2-linux-uclibc-gccbug
nios2-linux-ar nios2-linux-size nios2-linux-uclibc-gcov
nios2-linux-as nios2-linux-strings nios2-linux-uclibc-gprof
nios2-linux-c++filt nios2-linux-strip nios2-linux-uclibc-ld
nios2-linux-cpp nios2-linux-uclibc-addr2line nios2-linux-uclibc-ld.real
nios2-linux-gcc nios2-linux-uclibc-ar nios2-linux-uclibc-nm
nios2-linux-gccbug nios2-linux-uclibc-as nios2-linux-uclibc-objcopy
nios2-linux-gcov nios2-linux-uclibc-c++filt nios2-linux-uclibc-objdump
nios2-linux-gprof nios2-linux-uclibc-cc nios2-linux-uclibc-ranlib
nios2-linux-ld nios2-linux-uclibc-cpp nios2-linux-uclibc-readelf
nios2-linux-nm nios2-linux-uclibc-elf2flt nios2-linux-uclibc-size
nios2-linux-objcopy nios2-linux-uclibc-flthdr nios2-linux-uclibc-strings
nios2-linux-objdump nios2-linux-uclibc-gcc nios2-linux-uclibc-strip
nios2-linux-ranlib nios2-linux-uclibc-gcc-3.4.6
thx for help.
From spacelis at gmail.com Sat Jul 22 01:26:51 2006
From: spacelis at gmail.com (SpaceLi)
Date: Sat, 22 Jul 2006 16:26:51 +0800
Subject: [Live-devel] Is there any C edition of Live Media
Message-ID: <4f8ff5c10607220126y1432b9a9p8dc44dbdf278f090@mail.gmail.com>
I want port Live Media to uClinux, but C++ language seems not
supported. So could somebody send me the C edition of Live Media.thx
From kenneth.tu at ebell.com.tw Sat Jul 22 02:16:54 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Sat, 22 Jul 2006 17:16:54 +0800
Subject: [Live-devel] Could I Stream MPEG4 to Cell Phon?
Message-ID: <005a01c6ad6f$93952cc0$6501a8c0@kenneth>
Dear All.
I try to use testOnDemandRTSPServer streaming server streaming m4v file to cell phone via GPRS.
And then I try to take the stream by PacketVideo.(Sony Erricison P910i)
But It told me "MDS server can not access".
Did I do it right???
Thanks all
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From dsmurl at yahoo.com Sat Jul 22 14:48:35 2006
From: dsmurl at yahoo.com (Sam b)
Date: Sat, 22 Jul 2006 14:48:35 -0700 (PDT)
Subject: [Live-devel] Compiling darwin injector example
In-Reply-To: <9866ce4f0607211937l67cd9bedof10fb518012d8697@mail.gmail.com>
Message-ID: <20060722214835.18885.qmail@web38403.mail.mud.yahoo.com>
Ok. Thanks I was compiling them in the wrong order.
Even after I compiled it thought the makefile still
pointed towards needing basicusageenvironment.hh in
the livemedia/include directory, which it is not. So
I moved them in there and all the testprograms
compiled correctly then. Thanks for the hint.
Do you happen to know what direction I should take on
modifying the Darwin injector to accept a hinted mp4
stream? Or is there something that already does this?
- Thanks,
Sam
--- Scott Hays wrote:
> Have you compiled '../groupsock/libgroupsock.lib'?
> Scott
> > _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
__________________________________________________
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Tired of spam? Yahoo! Mail has the best spam protection around
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From finlayson at live555.com Sat Jul 22 16:31:17 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 22 Jul 2006 16:31:17 -0700
Subject: [Live-devel] Can I run LIVE.COM Streaming Media on my board of
Nios with uClinux?
In-Reply-To: <4f8ff5c10607212205j32324463x250a5b4ad4beedbf@mail.gmail.com>
References: <4f8ff5c10607212205j32324463x250a5b4ad4beedbf@mail.gmail.com>
Message-ID:
>I have searched the miling list on live555.com, but there is no hint
>for nios2. How can I port it to nios2.
Is this board running Linux? (It appears so, based on the compiler
tool names that you listed.) If so, then you should be able to
create your own "config.nios2" file (e.g., using the existing
"config.armlinux" file as a model), and then run "genMakefiles
nios2", and then "make".
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Sat Jul 22 16:43:20 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 22 Jul 2006 16:43:20 -0700
Subject: [Live-devel] Compiling darwin injector example
In-Reply-To: <20060722214835.18885.qmail@web38403.mail.mud.yahoo.com>
References: <20060722214835.18885.qmail@web38403.mail.mud.yahoo.com>
Message-ID:
>Ok. Thanks I was compiling them in the wrong order.
>Even after I compiled it thought the makefile still
>pointed towards needing basicusageenvironment.hh in
>the livemedia/include directory, which it is not.
The include file "BasicUsageEnvironment.hh" should be in the
"BasicUsageEnvironment/include" directory, *not* the
"liveMedia/include" directory. If you are building the "testProgs"
(or any other application), then you should be including both of
those directories (as well as "groupsock/include" and
"UsageEnvironment/include"). If you build the Makefiles properly,
then this should happen automatically.
>
>Do you happen to know what direction I should take on
>modifying the Darwin injector to accept a hinted mp4
>stream?
The "LIVE555 Streaming Media" code doesn't do anything with 'hinting'
- that's a hack that's used only by the Darwin Streaming Server
itself (when streaming from local files). I think you're really
asking how to demultiplex a MPEG-4-format file. This is something
that we don't yet support.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Sat Jul 22 16:44:55 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 22 Jul 2006 16:44:55 -0700
Subject: [Live-devel] Could I Stream MPEG4 to Cell Phon?
In-Reply-To: <005a01c6ad6f$93952cc0$6501a8c0@kenneth>
References: <005a01c6ad6f$93952cc0$6501a8c0@kenneth>
Message-ID:
>Dear All.
>I try to use testOnDemandRTSPServer streaming server streaming m4v
>file to cell phone via GPRS.
>And then I try to take the stream by PacketVideo.(Sony Erricison P910i)
>But It told me "MDS server can not access".
>Did I do it right???
Most likely, the problem is that your cellular phone provider does
not provide access to RTSP/RTP (or any other) stream that comes from
outside its network. Many cell phone providers are lame like this.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From finlayson at live555.com Sat Jul 22 17:04:24 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Sat, 22 Jul 2006 17:04:24 -0700
Subject: [Live-devel] Is there any C edition of Live Media
In-Reply-To: <4f8ff5c10607220126y1432b9a9p8dc44dbdf278f090@mail.gmail.com>
References: <4f8ff5c10607220126y1432b9a9p8dc44dbdf278f090@mail.gmail.com>
Message-ID:
>I want port Live Media to uClinux, but C++ language seems not
>supported
I don't believe this. Note that the GNU compiler (gcc) automatically
includes C++ support, so if you have GCC, you should have G++ also.
>. So could somebody send me the C edition of Live Media
Sorry there is none - nor will there ever be one. This is the 21st
Century. There is no longer any excuse for anyone supporting C, but
not C++.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From spacelis at gmail.com Sat Jul 22 22:42:40 2006
From: spacelis at gmail.com (SpaceLi)
Date: Sun, 23 Jul 2006 13:42:40 +0800
Subject: [Live-devel] Is there any C edition of Live Media
In-Reply-To:
References: <4f8ff5c10607220126y1432b9a9p8dc44dbdf278f090@mail.gmail.com>
Message-ID: <4f8ff5c10607222242i672a3d18wd4346d6e1e0412cb@mail.gmail.com>
Thank you for reply, but it still doesn't work.
I modified the file config.armlinux as
CROSSCOMPILE = nios2-linux
and then genMakefile and make.
but, it said that "C++ compiler not installed on this system"
and nios2-linux doesn't include g++ from nios2wiki, and there isn't
either in which I downloaded from
http://tinderbox.x86.dev.gentoo.org/cross-x86/
It seems very difficult to me to port it to nios2 with uclinux.
Could you give me some advice?
But nios2 ide 5.1 on winxp has g++ compiler, I don't know wether live
will work with it, because there is some thing with the stdlibc++, it
telled me that _Atomic_word doesn't name a type. I checked the include
path, seemed correct.
Should I change the lib?
This is the directories:
? iomanip
? ios
? iosfwd
? iostream
? istream
? iterator
????backward
? iomanip.h
? iostream.h
? istream.h
? ostream.h
? strstream
????bits
? allocator.h
? atomicity.h
? atomicity.h.bak
? basic_string.h
?
????debug
? debug.h
????nios2-elf
????bits
atomic_word.h
basic_file.h
c++allocator.h
c++config.h
c++io.h
c++locale.h
codecvt_specializations.h
ctype_base.h
ctype_inline.h
ctype_noninline.h
gthr-default.h
gthr-posix.h
gthr-single.h
gthr.h
messages_members.h
os_defines.h
time_members.h
_____________________________
#nios2-elf/bits/atomic_word.h
#ifndef _GLIBCXX_ATOMIC_WORD_H
#define _GLIBCXX_ATOMIC_WORD_H 1
typedef int _Atomic_word;
#endif
+++++++++++++++++++++++++++
#atomicity.h
#ifndef _GLIBCXX_ATOMICITY_H
#define _GLIBCXX_ATOMICITY_H 1
#include
namespace __gnu_cxx
{
_Atomic_word
__attribute__ ((__unused__))
__exchange_and_add(volatile _Atomic_word* __mem, int __val);
void
__attribute__ ((__unused__))
__atomic_add(volatile _Atomic_word* __mem, int __val);
} // namespace __gnu_cxx
#endif
++++++++++++++++++++++++++++++++
#basic_string.h
#ifndef _BASIC_STRING_H
#define _BASIC_STRING_H 1
#pragma GCC system_header
#include
#include
namespace std
{
template
class basic_string
{
// Types:
public:
typedef _Traits traits_type;
typedef typename _Traits::char_type value_type;
typedef _Alloc allocator_type;
typedef typename _Alloc::size_type size_type;
typedef typename _Alloc::difference_type difference_type;
typedef typename _Alloc::reference reference;
typedef typename _Alloc::const_reference const_reference;
typedef typename _Alloc::pointer pointer;
typedef typename _Alloc::const_pointer const_pointer;
typedef __gnu_cxx::__normal_iterator iterator;
typedef __gnu_cxx::__normal_iterator const_iterator;
typedef std::reverse_iterator const_reverse_iterator;
typedef std::reverse_iterator reverse_iterator;
private:
struct _Rep_base
{
size_type _M_length;
size_type _M_capacity;
_Atomic_word _M_refcount;
//int _M_refcount;
};
#.................
_____________________________
the error messages are:
nios2-elf-c++ -c -Iinclude -I../UsageEnvironment/include -I. -I/cygdrive/c/alter
a/kits/nios2_51/bin/eclipse/plugins/com.microtronix.nios2linux.uClibc_1.4.0/incl
ude -O2 -DSOCKLEN_T=socklen_t -D_LARGEFILE_SOURCE=1 -Wall -Wno-deprecated -DBSD
=1 -fexceptions Groupsock.cpp
In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
ios_base.h:45,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
9,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
ard/strstream:53,
from Groupsock.cpp:30:
/cygdrive/c/altera/kits/nios2_51/bin/eclipse/plugins/com.microtronix.nios2linux.
uClibc_1.4.0/include/bits/atomicity.h:25:2: warning: #warning stub atomicity fun
ctions are not really atomic
In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/strin
g:53,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
locale_classes.h:47,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
ios_base.h:47,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
9,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
ard/strstream:53,
from Groupsock.cpp:30:
/cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:147: er
ror: `_Atomic_word' does not name a type
/cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h: In mem
ber function `void std::basic_string<_CharT, _Traits, _Alloc>::_Rep::_M_dispose(
const _Alloc&)':
/cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:215: er
ror: `__exchange_and_add' is not a member of `__gnu_cxx'
/cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h: In mem
ber function `_CharT* std::basic_string<_CharT, _Traits, _Alloc>::_Rep::_M_refco
py()':
/cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:226: er
ror: `__atomic_add' is not a member of `__gnu_cxx'
In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
ios_base.h:47,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
9,
from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
-pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
ard/strstream:53,
On 7/23/06, Ross Finlayson wrote:
> >I want port Live Media to uClinux, but C++ language seems not
> >supported
>
> I don't believe this. Note that the GNU compiler (gcc) automatically
> includes C++ support, so if you have GCC, you should have G++ also.
>
> >. So could somebody send me the C edition of Live Media
>
> Sorry there is none - nor will there ever be one. This is the 21st
> Century. There is no longer any excuse for anyone supporting C, but
> not C++.
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From spacelis at gmail.com Sat Jul 22 23:05:42 2006
From: spacelis at gmail.com (SpaceLi)
Date: Sun, 23 Jul 2006 14:05:42 +0800
Subject: [Live-devel] Is there any C edition of Live Media
In-Reply-To: <4f8ff5c10607222242i672a3d18wd4346d6e1e0412cb@mail.gmail.com>
References: <4f8ff5c10607220126y1432b9a9p8dc44dbdf278f090@mail.gmail.com>
<4f8ff5c10607222242i672a3d18wd4346d6e1e0412cb@mail.gmail.com>
Message-ID: <4f8ff5c10607222305w356528b0u21a87a414e9ecba5@mail.gmail.com>
By the way,can I strip the muticast feature, becase of the error in
groupsock.cpp, and my project is dedicate to remote vod. Is there any
solution?
On 7/23/06, SpaceLi wrote:
> Thank you for reply, but it still doesn't work.
>
> I modified the file config.armlinux as
> CROSSCOMPILE = nios2-linux
> and then genMakefile and make.
> but, it said that "C++ compiler not installed on this system"
>
> and nios2-linux doesn't include g++ from nios2wiki, and there isn't
> either in which I downloaded from
> http://tinderbox.x86.dev.gentoo.org/cross-x86/
> It seems very difficult to me to port it to nios2 with uclinux.
> Could you give me some advice?
>
> But nios2 ide 5.1 on winxp has g++ compiler, I don't know wether live
> will work with it, because there is some thing with the stdlibc++, it
> telled me that _Atomic_word doesn't name a type. I checked the include
> path, seemed correct.
> Should I change the lib?
>
>
>
> This is the directories:
> ? iomanip
> ? ios
> ? iosfwd
> ? iostream
> ? istream
> ? iterator
> ????backward
> ? iomanip.h
> ? iostream.h
> ? istream.h
> ? ostream.h
> ? strstream
> ????bits
> ? allocator.h
> ? atomicity.h
> ? atomicity.h.bak
> ? basic_string.h
> ?
> ????debug
> ? debug.h
> ????nios2-elf
> ????bits
> atomic_word.h
> basic_file.h
> c++allocator.h
> c++config.h
> c++io.h
> c++locale.h
> codecvt_specializations.h
> ctype_base.h
> ctype_inline.h
> ctype_noninline.h
> gthr-default.h
> gthr-posix.h
> gthr-single.h
> gthr.h
> messages_members.h
> os_defines.h
> time_members.h
> _____________________________
>
> #nios2-elf/bits/atomic_word.h
> #ifndef _GLIBCXX_ATOMIC_WORD_H
> #define _GLIBCXX_ATOMIC_WORD_H 1
>
>
> typedef int _Atomic_word;
> #endif
> +++++++++++++++++++++++++++
> #atomicity.h
> #ifndef _GLIBCXX_ATOMICITY_H
> #define _GLIBCXX_ATOMICITY_H 1
>
> #include
> namespace __gnu_cxx
> {
> _Atomic_word
> __attribute__ ((__unused__))
> __exchange_and_add(volatile _Atomic_word* __mem, int __val);
>
> void
> __attribute__ ((__unused__))
> __atomic_add(volatile _Atomic_word* __mem, int __val);
> } // namespace __gnu_cxx
>
> #endif
> ++++++++++++++++++++++++++++++++
> #basic_string.h
> #ifndef _BASIC_STRING_H
> #define _BASIC_STRING_H 1
>
> #pragma GCC system_header
>
> #include
> #include
>
> namespace std
> {
> template
> class basic_string
> {
> // Types:
> public:
> typedef _Traits traits_type;
> typedef typename _Traits::char_type value_type;
> typedef _Alloc allocator_type;
> typedef typename _Alloc::size_type size_type;
> typedef typename _Alloc::difference_type difference_type;
> typedef typename _Alloc::reference reference;
> typedef typename _Alloc::const_reference const_reference;
> typedef typename _Alloc::pointer pointer;
> typedef typename _Alloc::const_pointer const_pointer;
> typedef __gnu_cxx::__normal_iterator iterator;
> typedef __gnu_cxx::__normal_iterator basic_string> const_iterator;
>
> typedef std::reverse_iterator const_reverse_iterator;
> typedef std::reverse_iterator reverse_iterator;
> private:
> struct _Rep_base
> {
> size_type _M_length;
> size_type _M_capacity;
> _Atomic_word _M_refcount;
> //int _M_refcount;
> };
> #.................
> _____________________________
> the error messages are:
>
> nios2-elf-c++ -c -Iinclude -I../UsageEnvironment/include -I. -I/cygdrive/c/alter
> a/kits/nios2_51/bin/eclipse/plugins/com.microtronix.nios2linux.uClibc_1.4.0/incl
> ude -O2 -DSOCKLEN_T=socklen_t -D_LARGEFILE_SOURCE=1 -Wall -Wno-deprecated -DBSD
> =1 -fexceptions Groupsock.cpp
> In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
> ios_base.h:45,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
> 9,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
> ard/strstream:53,
> from Groupsock.cpp:30:
> /cygdrive/c/altera/kits/nios2_51/bin/eclipse/plugins/com.microtronix.nios2linux.
> uClibc_1.4.0/include/bits/atomicity.h:25:2: warning: #warning stub atomicity fun
> ctions are not really atomic
> In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/strin
> g:53,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
> locale_classes.h:47,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
> ios_base.h:47,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
> 9,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
> ard/strstream:53,
> from Groupsock.cpp:30:
> /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
> gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:147: er
> ror: `_Atomic_word' does not name a type
> /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
> gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h: In mem
> ber function `void std::basic_string<_CharT, _Traits, _Alloc>::_Rep::_M_dispose(
> const _Alloc&)':
> /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
> gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:215: er
> ror: `__exchange_and_add' is not a member of `__gnu_cxx'
> /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
> gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h: In mem
> ber function `_CharT* std::basic_string<_CharT, _Traits, _Alloc>::_Rep::_M_refco
> py()':
> /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686-pc-cygwin/bin/../lib/
> gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/basic_string.h:226: er
> ror: `__atomic_add' is not a member of `__gnu_cxx'
> In file included from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/bits/
> ios_base.h:47,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/ios:4
> 9,
> from /cygdrive/c/altera/kits/nios2_51/bin/nios2-gnutools/H-i686
> -pc-cygwin/bin/../lib/gcc/nios2-elf/3.4.1/../../../../../include/c++/3.4.1/backw
> ard/strstream:53,
>
> On 7/23/06, Ross Finlayson wrote:
> > >I want port Live Media to uClinux, but C++ language seems not
> > >supported
> >
> > I don't believe this. Note that the GNU compiler (gcc) automatically
> > includes C++ support, so if you have GCC, you should have G++ also.
> >
> > >. So could somebody send me the C edition of Live Media
> >
> > Sorry there is none - nor will there ever be one. This is the 21st
> > Century. There is no longer any excuse for anyone supporting C, but
> > not C++.
> > --
> >
> > Ross Finlayson
> > Live Networks, Inc.
> > http://www.live555.com/
> > _______________________________________________
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> >
>
From weymar at ibr.cs.tu-bs.de Sun Jul 23 07:05:29 2006
From: weymar at ibr.cs.tu-bs.de (Rodrigo Weymar)
Date: Sun, 23 Jul 2006 16:05:29 +0200 (CEST)
Subject: [Live-devel] calling Live555 libraries from a C app under Embedded
Visual C++ 4.0
In-Reply-To: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
Message-ID: <55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
Hi all,
probably this is not the right mailing list to ask, since my question is
not directly concerning the Live555 libraries. But maybe someone can help
me or give me an advice.
I am trying to integrate the Live555 libs (I am interested in the RTSP
support provided by these libs) into an C app, more exactly a streaming
player for Pocket PC/WinCE420, which is written in C.
I use Embedded Visual C++ 4.0 as IDE.
I don't get any compilation errors, since from the C source code I am
using the following preprocessor directive:
#ifdef __cplusplus
extern "C" {
here I put the Live555 C++ classes
}
#endif
What happens is that, when I run the app, the Live555 classes don't take
any effect. That is, the Embedded Visual C++ C/C++ compiler seems to just
ignore what is in between the preprocessor directives. It seems to have
not linked the C++ classes to the C objs.
The document in [1] says that the C++ runtime libraries should be explicitly
linked to the app. The problem is that I can not figure out how Embedded
Visual C++ manages that. I was not able to find any option in "Project
Settings" or in "Tools -> Options" concerning that.
I also googled for solutions, but was not able to find one.
Would someone have previous experience with that? Could someone give me
help, please ?
Thanks a lot!
Rodrigo
[1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
From arcon-media at gmx.net Sun Jul 23 15:58:17 2006
From: arcon-media at gmx.net (arcon media)
Date: Mon, 24 Jul 2006 00:58:17 +0200
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
Message-ID:
> Perhaps. But it would be much better (and probably simpler) to use our
own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an encoder.
Hello,
How is this meant? Can you give me more information to this? It is very
important for me. I do not have any right idea for a good solution till now
yet. Till now, I only had worked with the VLC. I did not have any problems
with that. Unfortunately, the VLC does not support the demanded format of
the helix server.
I would be very pleased about good information.
Brian
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From sdhays.neon.com.tw at gmail.com Sun Jul 23 17:51:00 2006
From: sdhays.neon.com.tw at gmail.com (Scott Hays)
Date: Mon, 24 Jul 2006 08:51:00 +0800
Subject: [Live-devel] RTP to RTCP for Helix
In-Reply-To:
References:
Message-ID: <9866ce4f0607231751j7878d2bub25625acf95c95c7@mail.gmail.com>
Have you looked at MPEG4IP (http://mpeg4ip.sourceforge.net)?
I haven't used it for live streaming, but it's main focus seems to be
providing live streaming and it has
integration with FFMPEG. I don't know anything more about it's
streaming support since I only use some
of its utilities and libraries for other purposes.
Scott
On 7/24/06, arcon media wrote:
>
> > Perhaps. But it would be much better (and probably simpler) to use our
> own RTSP/RTP server implementation, with FFMPEG (e.g.) used as an encoder.
>
> Hello,
>
> How is this meant? Can you give me more information to this? It is very
> important for me. I do not have any right idea for a good solution till now
> yet. Till now, I only had worked with the VLC. I did not have any problems
> with that. Unfortunately, the VLC does not support the demanded format of
> the helix server.
>
> I would be very pleased about good information.
>
>
>
> Brian
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
-------------- next part --------------
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From junker at rbg.informatik.tu-darmstadt.de Mon Jul 24 00:05:09 2006
From: junker at rbg.informatik.tu-darmstadt.de (Bertram Junker)
Date: Mon, 24 Jul 2006 09:05:09 +0200
Subject: [Live-devel] calling Live555 libraries from a C app under
Embedded Visual C++ 4.0
In-Reply-To: <55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
<55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
Message-ID: <44C47125.3080106@rbg.informatik.tu-darmstadt.de>
Hi Rodrigo,
i have written an SIP phone with RTP Stack from LIVE555 for Pocket PC /
Windows CE with EVC++ 4.0. Unfortunately I'm not able to solve your
problem but I can ensure you, that LIVE555 works fine under EVC++.
Further, I remember me fine, I have not set up any preprocessor directive.
Best regards,
Bertram
> Hi all,
>
> probably this is not the right mailing list to ask, since my question is
> not directly concerning the Live555 libraries. But maybe someone can help
> me or give me an advice.
>
> I am trying to integrate the Live555 libs (I am interested in the RTSP
> support provided by these libs) into an C app, more exactly a streaming
> player for Pocket PC/WinCE420, which is written in C.
>
> I use Embedded Visual C++ 4.0 as IDE.
>
> I don't get any compilation errors, since from the C source code I am
> using the following preprocessor directive:
>
> #ifdef __cplusplus
> extern "C" {
>
> here I put the Live555 C++ classes
>
> }
> #endif
>
>
> What happens is that, when I run the app, the Live555 classes don't take
> any effect. That is, the Embedded Visual C++ C/C++ compiler seems to just
> ignore what is in between the preprocessor directives. It seems to have
> not linked the C++ classes to the C objs.
>
> The document in [1] says that the C++ runtime libraries should be explicitly
> linked to the app. The problem is that I can not figure out how Embedded
> Visual C++ manages that. I was not able to find any option in "Project
> Settings" or in "Tools -> Options" concerning that.
>
> I also googled for solutions, but was not able to find one.
>
> Would someone have previous experience with that? Could someone give me
> help, please ?
>
>
> Thanks a lot!
>
> Rodrigo
>
>
> [1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From tjc103 at ecs.soton.ac.uk Mon Jul 24 01:15:19 2006
From: tjc103 at ecs.soton.ac.uk (tjc103 at ecs.soton.ac.uk)
Date: Mon, 24 Jul 2006 09:15:19 +0100
Subject: [Live-devel] calling Live555 libraries from a C app under
Embedded Visual C++ 4.0
In-Reply-To: <55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
<55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
Message-ID: <1153728919.44c48197e1261@webmail.soton.ac.uk>
I managed to do it under Linux using GCC and G++ by having an extra C file
that calls the C++ functions, adding the line :
extern int cpp_function_i_want(void);
after all of the includes.
Then in a seperate cpp file I had the C++ code required. I then changed the
function description for the functions I wanted C to have access to :
extern "C" int cpp_function_i_want(){
This worked for everything I needed to do, I hope you have simliar success.
I also found this url usefull:
http://geneura.ugr.es/~jmerelo/c++-faq/mixing-c-and-cpp.html
Quoting Rodrigo Weymar :
> Hi all,
>
> probably this is not the right mailing list to ask, since my question is
> not directly concerning the Live555 libraries. But maybe someone can help
> me or give me an advice.
>
> I am trying to integrate the Live555 libs (I am interested in the RTSP
> support provided by these libs) into an C app, more exactly a streaming
> player for Pocket PC/WinCE420, which is written in C.
>
> I use Embedded Visual C++ 4.0 as IDE.
>
> I don't get any compilation errors, since from the C source code I am
> using the following preprocessor directive:
>
> #ifdef __cplusplus
> extern "C" {
>
> here I put the Live555 C++ classes
>
> }
> #endif
>
>
> What happens is that, when I run the app, the Live555 classes don't take
> any effect. That is, the Embedded Visual C++ C/C++ compiler seems to
> just
> ignore what is in between the preprocessor directives. It seems to have
> not linked the C++ classes to the C objs.
>
> The document in [1] says that the C++ runtime libraries should be
> explicitly
> linked to the app. The problem is that I can not figure out how Embedded
> Visual C++ manages that. I was not able to find any option in "Project
> Settings" or in "Tools -> Options" concerning that.
>
> I also googled for solutions, but was not able to find one.
>
> Would someone have previous experience with that? Could someone give me
> help, please ?
>
>
> Thanks a lot!
>
> Rodrigo
>
>
> [1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From ymreddy at ssdi.sharp.co.in Mon Jul 24 03:55:33 2006
From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT))
Date: Mon, 24 Jul 2006 16:25:33 +0530
Subject: [Live-devel] Sending BYE Packet...
Message-ID: <7FB4685EA93D014C8E30AA087B66E7520267BBA4@ssdimailsrvnt01.ssdi.sharp.co.in>
Hi Everybody,
Anybody tried sending the BYE packet from LIVE codebase. I am trying to send
the BYE packet when SSRC collision happens. I am trying this from
RTPSource.cpp file. How do I access the SendBYE() function defined RTCP.h. I
made the following statement "friend class RTPReceptionStatsDB;" in RTCP.h
file inorder to access SendBYE() function in RTPReceptionStatsDB() class.
I want to send the BYE packet with the RTCP instance already created in
main().
Any pointers for this solution, please send it.
Thanks and Regards
Y. Mallikharjuna Reddy
From shalom.shushan at gmail.com Mon Jul 24 04:53:45 2006
From: shalom.shushan at gmail.com (shalom shushan)
Date: Mon, 24 Jul 2006 13:53:45 +0200
Subject: [Live-devel] G.711 STREAMER
Message-ID:
hi,
I'm using the Live555 package and want to write G.711 audio streamer.
Is there anyone that already got it?
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From ymreddy at ssdi.sharp.co.in Mon Jul 24 08:13:12 2006
From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT))
Date: Mon, 24 Jul 2006 20:43:12 +0530
Subject: [Live-devel] Multicast support...
Message-ID: <7FB4685EA93D014C8E30AA087B66E7520267BBA6@ssdimailsrvnt01.ssdi.sharp.co.in>
Hi Everybody,
I am trying multicast the MPEG2 data using the testMPEG1or2VideoStreamer.cpp
file. I am using the multicast address as 239.255.42.42. I am getting the
following error:
20:04:32 Groupsock(920: 239.255.42.42, 8888, 1): failed to join group:
setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error
I am using Windows 2000 SP4. Any clues on how to solve this.
Thanks and Regards
Y. Mallikharjuna Reddy
From nagyz at nefty.hu Mon Jul 24 08:50:53 2006
From: nagyz at nefty.hu (Zoltan NAGY)
Date: Mon, 24 Jul 2006 17:50:53 +0200
Subject: [Live-devel] h263 support?
Message-ID: <44C4EC5D.6080600@nefty.hu>
Hello!
Is there h263 support in live.com? I dont see such rtsp server
inside testProgs...
Thanks,
Z
From ymreddy at ssdi.sharp.co.in Mon Jul 24 09:25:29 2006
From: ymreddy at ssdi.sharp.co.in (Mallikharjuna Reddy (NAVT))
Date: Mon, 24 Jul 2006 21:55:29 +0530
Subject: [Live-devel] SRHandler/RRHandler
Message-ID: <7FB4685EA93D014C8E30AA087B66E7520267BBA7@ssdimailsrvnt01.ssdi.sharp.co.in>
Hi Everybody,
I am trying to set the SR handler and RR handler functions in the main()
i.e. testMPEG1or2VideoStreamer.cpp/testMPEG1or2VideoReceiver.cpp files. I am
using the code like this:
sessionState.rtcpInstance->setSRHandler(sessionState.rtcpInstance->addReport
,buf);
It throws the following error message:
****************************************************************************
***********************************
Compiling...
testMPEG1or2VideoReceiver.cpp
E:\RTP-Source\LIVE20J1\Live12\Live555\testMPEG1or2VideoReceiver.cpp(126) :
error C2664: 'setSRHandler' : cannot convert parameter 1 from 'void (void)'
to 'void (__cdecl *)(void *)'
None of the functions with this name in scope match the target type
Error executing cl.exe.
****************************************************************************
************************************
The addReport() function is same as in RTCP.cpp file. Its used for testing
the handler functionality.
Any clues on the above, please reply.
Thanks and Regards
Y. Mallikharjuna Reddy
From xcsmith at rockwellcollins.com Mon Jul 24 11:35:36 2006
From: xcsmith at rockwellcollins.com (xcsmith at rockwellcollins.com)
Date: Mon, 24 Jul 2006 13:35:36 -0500
Subject: [Live-devel] MPEG Multiplexing
Message-ID:
Hello,
I'm working with a device which records analog audio/video and streams
either PES or MPEG2 TS to my application. Does the library provide a way
for me to multiplex Elementary Streams back into Program streams? I'd
prefer to store my data with my video and sound in the same .mpg file.
Thanks,
~Medra
From finlayson at live555.com Mon Jul 24 17:13:48 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 24 Jul 2006 17:13:48 -0700
Subject: [Live-devel] G.711 STREAMER
In-Reply-To:
References:
Message-ID:
>hi,
>
>I'm using the Live555 package and want to write G.711 audio streamer.
>Is there anyone that already got it?
G.711 is just u-law (or a-law) audio. Note the
"testWAVAudioStreamer" demo application (that streams audio from a
WAV audio file). If you uncomment the line
#define CONVERT_TO_ULAW 1
in "testWAVAudioStreamer.cpp", you will get G.711 u-law audio RTP streaming.
To stream G.711 a-law audio, you (or someone else) would need to
write a new "aLawFromPCMAudio" filter, similar to the existing
"uLawFromPCMAudio" filter class.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Mon Jul 24 17:15:23 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 24 Jul 2006 17:15:23 -0700
Subject: [Live-devel] h263 support?
In-Reply-To: <44C4EC5D.6080600@nefty.hu>
References: <44C4EC5D.6080600@nefty.hu>
Message-ID:
>Hello!
>
>Is there h263 support in live.com?
Yes, the library supports H.263/RTP streaming (both sending and
receiving). (However, none of the demo applications support it,
because there is no standard file format for H.263 video.)
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Mon Jul 24 17:23:52 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 24 Jul 2006 17:23:52 -0700
Subject: [Live-devel] MPEG Multiplexing
In-Reply-To:
References:
Message-ID:
>Does the library provide a way
>for me to multiplex Elementary Streams back into Program streams?
Not at present, no.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Mon Jul 24 17:26:35 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 24 Jul 2006 17:26:35 -0700
Subject: [Live-devel] Multicast support...
In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520267BBA6@ssdimailsrvnt01.ssdi.sharp.co.in
>
References: <7FB4685EA93D014C8E30AA087B66E7520267BBA6@ssdimailsrvnt01.ssdi.sharp.co.in
>
Message-ID:
>Hi Everybody,
>
>I am trying multicast the MPEG2 data using the testMPEG1or2VideoStreamer.cpp
>file. I am using the multicast address as 239.255.42.42. I am getting the
>following error:
>
>20:04:32 Groupsock(920: 239.255.42.42, 8888, 1): failed to join group:
>setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error
>
>I am using Windows 2000 SP4. Any clues on how to solve this.
You probably don't have a multicast route (for 224.0.0/4) in your
routing tables. If you add one (pointing to a local network
interface), multicast will probably work.
Or you could upgrade to Windows XP, where multicast seems to work
properly 'out of the box'.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Mon Jul 24 17:38:46 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 24 Jul 2006 17:38:46 -0700
Subject: [Live-devel] SRHandler/RRHandler
In-Reply-To: <7FB4685EA93D014C8E30AA087B66E7520267BBA7@ssdimailsrvnt01.ssdi.sharp.co.in
>
References: <7FB4685EA93D014C8E30AA087B66E7520267BBA7@ssdimailsrvnt01.ssdi.sharp.co.in
>
Message-ID:
>Hi Everybody,
>
>I am trying to set the SR handler and RR handler functions in the main()
>i.e. testMPEG1or2VideoStreamer.cpp/testMPEG1or2VideoReceiver.cpp files. I am
>using the code like this:
>
>sessionState.rtcpInstance->setSRHandler(sessionState.rtcpInstance->addReport
>,buf);
No, the "handlerFunc" parameter must have the signature
void TaskFunc(void* clientData)
In particular, it can't be a (non-static) class member function.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From shalom.shushan at gmail.com Mon Jul 24 23:37:11 2006
From: shalom.shushan at gmail.com (shalom shushan)
Date: Tue, 25 Jul 2006 08:37:11 +0200
Subject: [Live-devel] G.711 STREAMER
In-Reply-To:
References:
Message-ID:
Thank you Ross.
On 7/25/06, Ross Finlayson wrote:
>
> >hi,
> >
> >I'm using the Live555 package and want to write G.711 audio streamer.
> >Is there anyone that already got it?
>
> G.711 is just u-law (or a-law) audio. Note the
> "testWAVAudioStreamer" demo application (that streams audio from a
> WAV audio file). If you uncomment the line
> #define CONVERT_TO_ULAW 1
> in "testWAVAudioStreamer.cpp", you will get G.711 u-law audio RTP
> streaming.
>
> To stream G.711 a-law audio, you (or someone else) would need to
> write a new "aLawFromPCMAudio" filter, similar to the existing
> "uLawFromPCMAudio" filter class.
> --
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
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From barounis at ceid.upatras.gr Tue Jul 25 09:34:55 2006
From: barounis at ceid.upatras.gr (barounis at ceid.upatras.gr)
Date: Tue, 25 Jul 2006 19:34:55 +0300
Subject: [Live-devel] Final Question
In-Reply-To: <7.0.1.0.1.20060524195000.01ec82d8@live555.com>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
Message-ID: <1153845295.44c6482fbe9f9@my.ceid.upatras.gr>
?????? ?????? ??? Ross Finlayson :
>
> >I would like to have the ability to stop and then play another file
> >during the
> >same session RTSPServer from the RTSP server side. This means that I have
> to
> >call the MediaSink::stopPlaying or the destructor ~Mediasink().
>
> Just the former. Because you want to keep the RTP session intact,
> you should *not* close the (RTP)Sink object.
>
> To change the input source for a running stream, you should do the
> following, in order:
>
> sink->stopPlaying();
> Medium::close(oldSource);
> create newSource
> sink->startPlaying(newSource, ...);
>
>
> Ross Finlayson
> Live Networks, Inc. (LIVE555.COM)
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
Hello Ross,
First I would like to thank you for all the help you have given to me so far
and I really appreciate it as I am almost near to finish my master thesis.
I have one last question to ask you concerning the plan of the code you have
given to me as I don't want to bother anymore..
Before the stopPlaying() I would like the RTSPServer to have the ability to
know the exact time or frame of the video that has been played so far. After
creating the new source and before the play() I would like the RTSPServer to
make a seek() and then continue the streaming from the time or the frame of the
new video file.
This is my last question....
Can you please tell me how am I going to implement this?
Thanks for everything
Best regards
----------------------------------------------------
This mail was sent through http://my.ceid.upatras.gr
From kgurganus at 650dialup.com Tue Jul 25 09:42:55 2006
From: kgurganus at 650dialup.com (Keith Gurganus)
Date: Tue, 25 Jul 2006 12:42:55 -0400
Subject: [Live-devel] Streaming High Data Rate Video with DarwinInjector
Message-ID:
Hi all,
I have a requirement to stream high datarate video in the 10 ? 20 megabits
/sec range using the DarwinInjector class. I?ve tried this and it appears
to stall occasionally and never hit the target frames per second of 30 fps.
I can play the movie directly from file, so it appears not to be a decode /
playback issue. Any suggestions?
Thanks,
- Keith
From finlayson at live555.com Tue Jul 25 12:20:33 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 25 Jul 2006 12:20:33 -0700
Subject: [Live-devel] Final Question
In-Reply-To: <1153845295.44c6482fbe9f9@my.ceid.upatras.gr>
References: <1143586422.4429be76673a7@my.ceid.upatras.gr>
<7.0.1.0.1.20060328150729.01f93568@live555.com>
<1144792970.443c278a9bc41@my.ceid.upatras.gr>
<7.0.1.0.1.20060411151024.01d638a0@live555.com>
<1146055961.444f6d19519e3@my.ceid.upatras.gr>
<1146132663.445098b73a8d8@my.ceid.upatras.gr>
<7.0.1.0.1.20060427053054.01f51830@live555.com>
<1146146329.4450ce19cb6d9@my.ceid.upatras.gr>
<7.0.1.0.1.20060427073327.01f7ed50@live555.com>
<1146923099.445ca85b050bc@my.ceid.upatras.gr>
<7.0.1.0.1.20060506224156.01f868d8@live555.com>
<1147875340.446b300cb6f38@my.ceid.upatras.gr>
<1148308140.4471caacbc138@my.ceid.upatras.gr>
<1148508296.4474d888245e1@my.ceid.upatras.gr>
<7.0.1.0.1.20060524195000.01ec82d8@live555.com>
<1153845295.44c6482fbe9f9@my.ceid.upatras.gr>
Message-ID:
>Before the stopPlaying() I would like the RTSPServer to have the ability to
>know the exact time or frame of the video that has been played so far
You will have to get this information from the "RTPSink" object. I
suggest modifying "MultiFramedRTPSink::setTimestamp()" to record the
"timestamp" parameter. This is will be the presentation time
corresponding the most recently-sent packet.
Apart from this - because I don't know much about your application -
I'm not going to be able to give you any more help here.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Tue Jul 25 12:30:51 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Tue, 25 Jul 2006 12:30:51 -0700
Subject: [Live-devel] Streaming High Data Rate Video with DarwinInjector
In-Reply-To:
References:
Message-ID:
>I have a requirement to stream high datarate video in the 10 - 20 megabits
>/sec range using the DarwinInjector class. I've tried this and it appears
>to stall occasionally and never hit the target frames per second of 30 fps.
>I can play the movie directly from file, so it appears not to be a decode /
>playback issue. Any suggestions?
This sounds like a networking/TCP issue between the DarwinInjector
application and the DSS. (Recall that injection into a DSS uses
RTP/RTCP-over-TCP.) Are these running on the same computer, or on
separate computers? If they're on separate computers, then perhaps
you're getting network congestion. If they're on the same computer,
then perhaps there's some OS TCP buffering parameter that you can
tweak??
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From kenneth.tu at ebell.com.tw Tue Jul 25 22:05:20 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Wed, 26 Jul 2006 13:05:20 +0800
Subject: [Live-devel] Destination Unreachable (Port unreachable)
Message-ID: <000f01c6b071$18364280$386e55d2@kenneth>
Dear all
I have used the testOnDemandRTSPServer to streaming m4v file.
I met a problem like this.
When some one try to access the RTSP Server and then he got "Bad request".
I trace the RTSP command I got this "DESCRIBE -> SETUP -> TEARDOWN".
And finally, I use Ethereal to get the packets.
I found this.
Protocol info
ICMP Destination Unreachable (Port unreachable)
Could someone tell me what wrong?
Thanks
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From finlayson at live555.com Wed Jul 26 02:32:58 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 26 Jul 2006 02:32:58 -0700
Subject: [Live-devel] Destination Unreachable (Port unreachable)
In-Reply-To: <000f01c6b071$18364280$386e55d2@kenneth>
References: <000f01c6b071$18364280$386e55d2@kenneth>
Message-ID:
>Dear all
>I have used the testOnDemandRTSPServer to streaming m4v file.
>I met a problem like this.
>When some one try to access the RTSP Server and then he got "Bad request".
>I trace the RTSP command I got this "DESCRIBE -> SETUP -> TEARDOWN".
>And finally, I use Ethereal to get the packets.
>I found this.
>
>Protocol info
>ICMP Destination Unreachable (Port unreachable)
>
>Could someone tell me what wrong?
My guess is that you have a firewall somewhere that's blocking that port.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From weymar at ibr.cs.tu-bs.de Wed Jul 26 02:51:03 2006
From: weymar at ibr.cs.tu-bs.de (Rodrigo Weymar)
Date: Wed, 26 Jul 2006 11:51:03 +0200 (CEST)
Subject: [Live-devel] calling Live555 libraries from a C app under
Embedded Visual C++ 4.0
In-Reply-To: <1153728919.44c48197e1261@webmail.soton.ac.uk>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
<55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
<1153728919.44c48197e1261@webmail.soton.ac.uk>
Message-ID: <49843.134.169.35.238.1153907463.squirrel@webmail.ibr.cs.tu-bs.de>
Hi,
thank you very much for your help.
regards,
Rodrigo
> I managed to do it under Linux using GCC and G++ by having an extra C file
> that calls the C++ functions, adding the line :
>
> extern int cpp_function_i_want(void);
>
> after all of the includes.
>
> Then in a seperate cpp file I had the C++ code required. I then changed
> the
> function description for the functions I wanted C to have access to :
>
> extern "C" int cpp_function_i_want(){
>
> This worked for everything I needed to do, I hope you have simliar
> success.
> I also found this url usefull:
>
> http://geneura.ugr.es/~jmerelo/c++-faq/mixing-c-and-cpp.html
>
>
>
>
> Quoting Rodrigo Weymar :
>
>> Hi all,
>>
>> probably this is not the right mailing list to ask, since my question is
>> not directly concerning the Live555 libraries. But maybe someone can
>> help
>> me or give me an advice.
>>
>> I am trying to integrate the Live555 libs (I am interested in the RTSP
>> support provided by these libs) into an C app, more exactly a streaming
>> player for Pocket PC/WinCE420, which is written in C.
>>
>> I use Embedded Visual C++ 4.0 as IDE.
>>
>> I don't get any compilation errors, since from the C source code I am
>> using the following preprocessor directive:
>>
>> #ifdef __cplusplus
>> extern "C" {
>>
>> here I put the Live555 C++ classes
>>
>> }
>> #endif
>>
>>
>> What happens is that, when I run the app, the Live555 classes don't take
>> any effect. That is, the Embedded Visual C++ C/C++ compiler seems to
>> just
>> ignore what is in between the preprocessor directives. It seems to have
>> not linked the C++ classes to the C objs.
>>
>> The document in [1] says that the C++ runtime libraries should be
>> explicitly
>> linked to the app. The problem is that I can not figure out how Embedded
>> Visual C++ manages that. I was not able to find any option in "Project
>> Settings" or in "Tools -> Options" concerning that.
>>
>> I also googled for solutions, but was not able to find one.
>>
>> Would someone have previous experience with that? Could someone give me
>> help, please ?
>>
>>
>> Thanks a lot!
>>
>> Rodrigo
>>
>>
>> [1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From weymar at ibr.cs.tu-bs.de Wed Jul 26 03:08:44 2006
From: weymar at ibr.cs.tu-bs.de (Rodrigo Weymar)
Date: Wed, 26 Jul 2006 12:08:44 +0200 (CEST)
Subject: [Live-devel] calling Live555 libraries from a C app under
Embedded Visual C++ 4.0
In-Reply-To: <44C47125.3080106@rbg.informatik.tu-darmstadt.de>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk>
<55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de>
<44C47125.3080106@rbg.informatik.tu-darmstadt.de>
Message-ID: <49884.134.169.35.238.1153908524.squirrel@webmail.ibr.cs.tu-bs.de>
Hi Bertram,
is your application written in C or C++ ? Were you able to mix C and C++
code and to declare C++ functions as extern "C" under EVC++ without using
#ifdef __cplusplus ?
If I try to use extern "C" under EVC++ without using #ifdef __cplusplus, I
get a
error C2059: syntax error : 'string'
as explained in http://www.kbalertz.com/kb_133070.aspx
regards,
Rodrigo
> Hi Rodrigo,
>
> i have written an SIP phone with RTP Stack from LIVE555 for Pocket PC /
> Windows CE with EVC++ 4.0. Unfortunately I'm not able to solve your
> problem but I can ensure you, that LIVE555 works fine under EVC++.
> Further, I remember me fine, I have not set up any preprocessor directive.
>
> Best regards,
>
> Bertram
>
>
>> Hi all,
>>
>> probably this is not the right mailing list to ask, since my question is
>> not directly concerning the Live555 libraries. But maybe someone can
>> help
>> me or give me an advice.
>>
>> I am trying to integrate the Live555 libs (I am interested in the RTSP
>> support provided by these libs) into an C app, more exactly a streaming
>> player for Pocket PC/WinCE420, which is written in C.
>>
>> I use Embedded Visual C++ 4.0 as IDE.
>>
>> I don't get any compilation errors, since from the C source code I am
>> using the following preprocessor directive:
>>
>> #ifdef __cplusplus
>> extern "C" {
>>
>> here I put the Live555 C++ classes
>>
>> }
>> #endif
>>
>>
>> What happens is that, when I run the app, the Live555 classes don't take
>> any effect. That is, the Embedded Visual C++ C/C++ compiler seems to
>> just
>> ignore what is in between the preprocessor directives. It seems to have
>> not linked the C++ classes to the C objs.
>>
>> The document in [1] says that the C++ runtime libraries should be
>> explicitly
>> linked to the app. The problem is that I can not figure out how Embedded
>> Visual C++ manages that. I was not able to find any option in "Project
>> Settings" or in "Tools -> Options" concerning that.
>>
>> I also googled for solutions, but was not able to find one.
>>
>> Would someone have previous experience with that? Could someone give me
>> help, please ?
>>
>>
>> Thanks a lot!
>>
>> Rodrigo
>>
>>
>> [1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
>>
>> _______________________________________________
>> live-devel mailing list
>> live-devel at lists.live555.com
>> http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From larissalucena at gmail.com Wed Jul 26 07:07:43 2006
From: larissalucena at gmail.com (Larissa Lucena)
Date: Wed, 26 Jul 2006 11:07:43 -0300
Subject: [Live-devel] JPEG video
Message-ID: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
Hi there,
I'm developing an application that I have to use videos like various JPEGs.
I'm using the live, but I don't know how to do it even so... could you help
me? Maybe an example code?
Thanks,
Larissa
--
"O maior prazer do inteligente ? bancar o idiota
diante de um idiota que banca o inteligente".
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From yossyd at nayos.com Wed Jul 26 08:12:43 2006
From: yossyd at nayos.com (Yossy Dreyfus)
Date: Wed, 26 Jul 2006 17:12:43 +0200
Subject: [Live-devel] G.711 STREAMER
In-Reply-To:
Message-ID: <000001c6b0c5$f24dd1f0$570a1f0a@nayos.local>
>hi,
>
>I'm using the Live555 package and want to write G.711 audio streamer.
>Is there anyone that already got it?
G.711 is just u-law (or a-law) audio. Note the
"testWAVAudioStreamer" demo application (that streams audio from a
WAV audio file). If you uncomment the line
#define CONVERT_TO_ULAW 1
in "testWAVAudioStreamer.cpp", you will get G.711 u-law audio RTP
streaming.
To stream G.711 a-law audio, you (or someone else) would need to
write a new "aLawFromPCMAudio" filter, similar to the existing
"uLawFromPCMAudio" filter class.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
Hello,
I have an a-law audio file, so I don't need the "aLawFromPCMAudio".
I have tried to send it with "testWAVAudioStreamer" but the player got
only noise. How can I fix it?
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From kenneth.tu at ebell.com.tw Wed Jul 26 07:51:12 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Wed, 26 Jul 2006 22:51:12 +0800
Subject: [Live-devel] Streaming over Http
Message-ID: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
Could Live Server stream video over HTTP??
Thanks
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From lorenooliveira at gmail.com Wed Jul 26 09:50:39 2006
From: lorenooliveira at gmail.com (Loreno Oliveira)
Date: Wed, 26 Jul 2006 13:50:39 -0300
Subject: [Live-devel] JPEG video
In-Reply-To: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
References: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
Message-ID:
Oq eh que tu precisa fazer? eh pra reproduzir ou transmitir via RTP?
On 7/26/06, Larissa Lucena wrote:
>
> Hi there,
>
> I'm developing an application that I have to use videos like various
> JPEGs. I'm using the live, but I don't know how to do it even so... could
> you help me? Maybe an example code?
>
> Thanks,
>
> Larissa
>
>
> --
> "O maior prazer do inteligente ? bancar o idiota
> diante de um idiota que banca o inteligente".
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
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From finlayson at live555.com Wed Jul 26 14:17:57 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 26 Jul 2006 14:17:57 -0700
Subject: [Live-devel] G.711 STREAMER
In-Reply-To: <000001c6b0c5$f24dd1f0$570a1f0a@nayos.local>
References: <000001c6b0c5$f24dd1f0$570a1f0a@nayos.local>
Message-ID:
>I have an a-law audio file, so I don't need the "aLawFromPCMAudio".
>I have tried to send it with "testWAVAudioStreamer" but the player
>got only noise. How can I fix it?
Is your file really a ".wav" file?? "testWAVAudioStreamer" is (as
its name suggests) specifically for ".wav" PCM files.
Streaming a-law audio over RTP is done the same way as streaming
u-law audio over RTP, except that the MIME subtype is "PCMA" (rather
than "PCMU"), and - for 8 kHz mono - the static payload type 8
(rather than 0). (See RFC 3551.)
You should be able to figure out from the existing code how to do this...
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From finlayson at live555.com Wed Jul 26 15:12:20 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 26 Jul 2006 15:12:20 -0700
Subject: [Live-devel] Streaming over Http
In-Reply-To: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
References: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
Message-ID:
>Could Live Server stream video over HTTP??
Are you referring to the "RTSP-over-RTP" hack that QuickTime Player
(and the Darwin Streaming Server) uses as an option in order to
tunnel RTSP/RTP/RTCP over a firewall? Or are you simply referring to
streaming video over a HTTP connection, without using RTSP/RTP/RTCP
at all?
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From kenneth.tu at ebell.com.tw Wed Jul 26 18:56:40 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Thu, 27 Jul 2006 09:56:40 +0800
Subject: [Live-devel] Streaming over Http
References: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
Message-ID: <001201c6b11f$e7b243c0$6401a8c0@kenneth>
Re: [Live-devel] Streaming over HttpThanks Ross
I just wanna stream video over a HTTP connection.
Because I met some situation bad for RTSP connection.
I try to use RealPlayer to access Live Streaming Server by DSL network and It got a "Bad Requset".
It seemed block by someone.
So I simply think If I could stream over HTTP it may be solved.
----- Original Message -----
From: Ross Finlayson
To: LIVE555 Streaming Media - development & use
Sent: Thursday, July 27, 2006 6:12 AM
Subject: Re: [Live-devel] Streaming over Http
Could Live Server stream video over HTTP??
Are you referring to the "RTSP-over-RTP" hack that QuickTime Player (and the Darwin Streaming Server) uses as an option in order to tunnel RTSP/RTP/RTCP over a firewall? Or are you simply referring to streaming video over a HTTP connection, without using RTSP/RTP/RTCP at all?
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
------------------------------------------------------------------------------
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
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From kenneth.tu at ebell.com.tw Wed Jul 26 19:01:35 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Thu, 27 Jul 2006 10:01:35 +0800
Subject: [Live-devel] Streaming over Http
References: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
Message-ID: <002a01c6b120$971edd50$6401a8c0@kenneth>
Re: [Live-devel] Streaming over HttpThanks Ross
I just wanna stream video over a HTTP connection.
Because I met some situation bad for RTSP connection.
I try to use RealPlayer to access Live Streaming Server by DSL network and It got a "Bad Requset".
It seemed block by someone.
So I simply think If I could stream over HTTP it may be solved.
----- Original Message -----
From: Ross Finlayson
To: LIVE555 Streaming Media - development & use
Sent: Thursday, July 27, 2006 6:12 AM
Subject: Re: [Live-devel] Streaming over Http
Could Live Server stream video over HTTP??
Are you referring to the "RTSP-over-RTP" hack that QuickTime Player (and the Darwin Streaming Server) uses as an option in order to tunnel RTSP/RTP/RTCP over a firewall? Or are you simply referring to streaming video over a HTTP connection, without using RTSP/RTP/RTCP at all?
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
------------------------------------------------------------------------------
_______________________________________________
live-devel mailing list
live-devel at lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel
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From finlayson at live555.com Wed Jul 26 18:58:35 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Wed, 26 Jul 2006 18:58:35 -0700
Subject: [Live-devel] Streaming over Http
In-Reply-To: <001201c6b11f$e7b243c0$6401a8c0@kenneth>
References: <001601c6b0c2$f1a53d40$6401a8c0@kenneth>
<001201c6b11f$e7b243c0$6401a8c0@kenneth>
Message-ID:
>I just wanna stream video over a HTTP connection.
>
>Because I met some situation bad for RTSP connection.
>
>I try to use RealPlayer to access Live Streaming Server by DSL
>network and It got a "Bad Requset".
RealPlayer's RTSP/RTP support is badly broken. Try VLC instead.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From kenneth.tu at ebell.com.tw Thu Jul 27 00:36:48 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Thu, 27 Jul 2006 15:36:48 +0800
Subject: [Live-devel] RTCP
Message-ID: <000d01c6b14f$6b83f2a0$6601a8c0@kenneth>
Dear all
I met a serious problem.
When I stream on dsl network it almost block all of the bandwidth.
Could I control the send frequency by somwhat?(like RTCP??)
Thanks a lot
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From nagyz at nefty.hu Thu Jul 27 00:33:52 2006
From: nagyz at nefty.hu (Zoltan NAGY)
Date: Thu, 27 Jul 2006 09:33:52 +0200
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
Message-ID: <44C86C60.9050108@nefty.hu>
Hello list!
I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
testH263VideoStreamer and testMPEG4VideoStreamer to stream
videos to my Nokia N70.
Has anybody got any suggestions? I'm welcome to anything..
The telephone sometimes sais "Unable to connect", sometimes
it sais "Loading" for ~20-30secs, then "Unable to open"...
I've tried to open the rtsp streams with VLC/MPlayer - unsuccessfully, also.
Anybody got that running?
Ross?
Any help would be greatly appriciated...
Thanks,
Zoltan
From finlayson at live555.com Thu Jul 27 00:55:30 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 27 Jul 2006 00:55:30 -0700
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
In-Reply-To: <44C86C60.9050108@nefty.hu>
References: <44C86C60.9050108@nefty.hu>
Message-ID:
>Hello list!
>
>I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
>testH263VideoStreamer and testMPEG4VideoStreamer to stream
>videos to my Nokia N70.
Most likely, the problem is that your cellular phone provider does
not provide access to any RTSP/RTP (or any other) stream that comes
from outside its network. Many (although not all) cell phone
providers are restrictive like this.
The way you can test this is by adding
#define DEBUG 1
to the start of "liveMedia/RTSPServer.cpp" and recompiling. This
will show whether or not any RTSP commands are reaching the server
from your client.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From nagyz at nefty.hu Thu Jul 27 01:13:21 2006
From: nagyz at nefty.hu (Zoltan NAGY)
Date: Thu, 27 Jul 2006 10:13:21 +0200
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
In-Reply-To:
References: <44C86C60.9050108@nefty.hu>
Message-ID: <44C875A1.3040606@nefty.hu>
Ross Finlayson wrote:
>> Hello list!
>>
>> I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
>> testH263VideoStreamer and testMPEG4VideoStreamer to stream
>> videos to my Nokia N70.
>>
>
> Most likely, the problem is that your cellular phone provider does
> not provide access to any RTSP/RTP (or any other) stream that comes
> from outside its network. Many (although not all) cell phone
> providers are restrictive like this.
>
>
ok, that's not the problem.
the telephone was able to connect to the server, but timed out.. at
least the
telephone said that.
the full log is available at http://nefty.hu/~nagyz/live-log.txt
ideas?
Zoltan
From kenneth.tu at ebell.com.tw Thu Jul 27 01:43:12 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Thu, 27 Jul 2006 16:43:12 +0800
Subject: [Live-devel] RTCP
Message-ID: <000c01c6b158$b2384b70$6401a8c0@kenneth>
Dear all
I met a serious problem.
When I stream on dsl network it almost block all of the bandwidth.
Could I control the send frequency by somwhat?(like RTCP??)
Thanks a lot
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From millallo at gmail.com Thu Jul 27 01:43:40 2006
From: millallo at gmail.com (Emiliano Parasassi)
Date: Thu, 27 Jul 2006 10:43:40 +0200
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
In-Reply-To: <44C875A1.3040606@nefty.hu>
References: <44C86C60.9050108@nefty.hu>
<44C875A1.3040606@nefty.hu>
Message-ID: <44C87CBC.8020803@gmail.com>
Zoltan NAGY wrote:
> Ross Finlayson wrote:
>>> Hello list!
>>>
>>> I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
>>> testH263VideoStreamer and testMPEG4VideoStreamer to stream
>>> videos to my Nokia N70.
>>>
>> Most likely, the problem is that your cellular phone provider does
>> not provide access to any RTSP/RTP (or any other) stream that comes
>> from outside its network. Many (although not all) cell phone
>> providers are restrictive like this.
>>
>>
> ok, that's not the problem.
> the telephone was able to connect to the server, but timed out.. at
> least the
> telephone said that.
>
> the full log is available at http://nefty.hu/~nagyz/live-log.txt
>
> ideas?
>
> Zoltan
>
Are you sure that RTP/RTCP packets sent by server arrive to client? Most
problably they are dropped by your provider. Without them nothing can be
played. RTSP(tcp) is used only to start and control the stream and seems to work,
but the true stream is in RTP(udp) packets, and RTCP(udp) is used to provide quality
information of the RTP flow.
Bye
Emiliano
From nagyz at nefty.hu Thu Jul 27 02:05:01 2006
From: nagyz at nefty.hu (Zoltan NAGY)
Date: Thu, 27 Jul 2006 11:05:01 +0200
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
In-Reply-To: <44C87CBC.8020803@gmail.com>
References: <44C86C60.9050108@nefty.hu>
<44C875A1.3040606@nefty.hu>
<44C87CBC.8020803@gmail.com>
Message-ID: <44C881BD.1060807@nefty.hu>
Emiliano Parasassi wrote:
> Zoltan NAGY wrote:
>
>> Ross Finlayson wrote:
>>
>>>> Hello list!
>>>>
>>>> I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
>>>> testH263VideoStreamer and testMPEG4VideoStreamer to stream
>>>> videos to my Nokia N70.
>>>>
>>>>
>>> Most likely, the problem is that your cellular phone provider does
>>> not provide access to any RTSP/RTP (or any other) stream that comes
>>> from outside its network. Many (although not all) cell phone
>>> providers are restrictive like this.
>>>
>>>
>>>
>> ok, that's not the problem.
>> the telephone was able to connect to the server, but timed out.. at
>> least the
>> telephone said that.
>>
>> the full log is available at http://nefty.hu/~nagyz/live-log.txt
>>
>> ideas?
>>
>> Zoltan
>>
>>
>
> Are you sure that RTP/RTCP packets sent by server arrive to client? Most
> problably they are dropped by your provider. Without them nothing can be
> played. RTSP(tcp) is used only to start and control the stream and seems to work,
> but the true stream is in RTP(udp) packets, and RTCP(udp) is used to provide quality
> information of the RTP flow.
>
well, I'm not sure.. and I'm not sure either how to check this..
any ideas?
if I call customer service, they'll laugh me in the face, I think, as
they'll
have no idea what RTP/RTCP is.. :)
Thanks,
Zoltan
From finlayson at live555.com Thu Jul 27 02:17:15 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 27 Jul 2006 02:17:15 -0700
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
In-Reply-To: <44C881BD.1060807@nefty.hu>
References: <44C86C60.9050108@nefty.hu>
<44C875A1.3040606@nefty.hu>
<44C87CBC.8020803@gmail.com> <44C881BD.1060807@nefty.hu>
Message-ID:
> > Are you sure that RTP/RTCP packets sent by server arrive to client? Most
>> problably they are dropped by your provider. Without them nothing can be
>> played. RTSP(tcp) is used only to start and control the stream and
>>seems to work,
>> but the true stream is in RTP(udp) packets, and RTCP(udp) is used
>>to provide quality
>> information of the RTP flow.
>>
>well, I'm not sure.. and I'm not sure either how to check this..
>any ideas?
Check whether "MultiFramedRTPSource::networkReadHandler()" ever gets
called. If it doesn't, then no RTP packets are arriving.
From bidibulle at operamail.com Thu Jul 27 02:40:44 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 27 Jul 2006 10:40:44 +0100
Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
Message-ID: <20060727094044.6EA1124516@ws5-3.us4.outblaze.com>
Try also to change the server port used by the test programs to be te default RTSP port (554). Most mobile providers use strict NAT/Firewall policies for packets coming from outside (like RTP/RTCP packets sent by streaming servers). However, if you use a standard port, your provider might use a special -more tolerant- policy to enable streaming (sometimes by adding an RTSP ALG module in the way)..
Be carefull, on a Linux platform, you need to be root to be able to open port 554 (all ports < 1024 by the way)
If it fails, then you have to ry another APN or another provider ...
David
> ----- Original Message -----
> From: "Zoltan NAGY"
> To: "LIVE555 Streaming Media - development & use"
> Subject: [Live-devel] h263/mpeg4 streaming to nokia phones?
> Date: Thu, 27 Jul 2006 09:33:52 +0200
>
>
> Hello list!
>
> I've been trying - unsuccessfully - to use testOnDemandRTSPServer,
> testH263VideoStreamer and testMPEG4VideoStreamer to stream
> videos to my Nokia N70.
>
> Has anybody got any suggestions? I'm welcome to anything..
>
> The telephone sometimes sais "Unable to connect", sometimes
> it sais "Loading" for ~20-30secs, then "Unable to open"...
>
> I've tried to open the rtsp streams with VLC/MPlayer - unsuccessfully, also.
>
> Anybody got that running?
> Ross?
>
> Any help would be greatly appriciated...
>
> Thanks,
>
> Zoltan
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 9 at http://www.opera.com
Powered by Outblaze
From larissalucena at gmail.com Thu Jul 27 04:30:49 2006
From: larissalucena at gmail.com (Larissa Lucena)
Date: Thu, 27 Jul 2006 08:30:49 -0300
Subject: [Live-devel] JPEG video
In-Reply-To:
References: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
Message-ID: <440165240607270430lc299512o9645f56d41448a80@mail.gmail.com>
P/ transmitir via RTP...
On 7/26/06, Loreno Oliveira wrote:
>
> Oq eh que tu precisa fazer? eh pra reproduzir ou transmitir via RTP?
>
>
>
> On 7/26/06, Larissa Lucena wrote:
>
> > Hi there,
>
> I'm developing an application that I have to use videos like various
> JPEGs. I'm using the live, but I don't know how to do it even so... could
> you help me? Maybe an example code?
>
> Thanks,
>
> Larissa
>
>
> --
> "O maior prazer do inteligente ? bancar o idiota
> diante de um idiota que banca o inteligente".
>
> _______________________________________________
>
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
>
>
--
"O maior prazer do inteligente ? bancar o idiota
diante de um idiota que banca o inteligente".
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From tjc103 at ecs.soton.ac.uk Thu Jul 27 08:55:19 2006
From: tjc103 at ecs.soton.ac.uk (tjc103 at ecs.soton.ac.uk)
Date: Thu, 27 Jul 2006 16:55:19 +0100
Subject: [Live-devel] Stream discovery
Message-ID: <1154015719.44c8e1e79effa@webmail.soton.ac.uk>
Hi
I understand that using RTSP it is possible to get a description of a known
streaming using DESCRIBE. For instance if I use the openRTSP client and
connect it to a stream rtsp://192.168.0.1:8080/testStream I'll get :
v=0
o=- 1154014890697385 1 IN IP4 130.141.6.75
s=Session streamed by "testOnDemandRTSPServer"
i=testStream
t=0 0
a=tool:LIVE555 Streaming Media v2006.07.03
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session streamed by "testOnDemandRTSPServer"
a=x-qt-text-inf:testStream
m=video 0 RTP/AVP 33
c=IN IP4 0.0.0.0
a=control:track1
But if I point it to just rtsp://192.168.0.1:8080/, I'll get nothing. Is
there a standard way of getting the server to list all the available
transport streams?
Thanks
Theo
From bidibulle at operamail.com Thu Jul 27 11:50:41 2006
From: bidibulle at operamail.com (David BERTRAND)
Date: Thu, 27 Jul 2006 19:50:41 +0100
Subject: [Live-devel] RTCPInstance and RTP timestamp
Message-ID: <20060727185041.292B543CC2@ws5-1.us4.outblaze.com>
Ross,
Thank yo ufor including this in next release. However, I would like to update slightly my patch. Actually, previous patch didn't prevent from sending the RTCP packet, just omitted the report inside. This lead to a source description RTCP packet, which IMO is not a good idea. Therefore , I now make my check (for a RTP packet being sent firstly) in sendReport() i.s.o addReport() :
void RTCPInstance::sendReport() {
if (fSink != NULL && !(fSink->haveComputedFirstTimestamp())) {
return;
}
#ifdef DEBUG
fprintf(stderr, "sending REPORT\n");
#endif
[...]
Sorry for publishing this fix in two times.
David
> ----- Original Message -----
> From: "Ross Finlayson"
> To: "LIVE555 Streaming Media - development & use"
> Subject: Re: [Live-devel] RTCPInstance and RTP timestamp
> Date: Wed, 19 Jul 2006 05:07:55 -0700
>
>
> > I recognize my problem is quite rare (if you stream from a file,
> > you can send RTP packets immediately, so RTCP packets will
> > probably be sent after the first RTP packets) and also very
> > difficult to resolve properly. IMO, the only acceptable way is to
> > send RTCP packets only if a RTP packet has already been sent for
> > the same track. This fix works for my particular problem and I
> > would like to propose it as patch for the library.
>
> That looks reasonable. It will be included in the next release of
> the software.
> --
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
--
_______________________________________________
Surf the Web in a faster, safer and easier way:
Download Opera 9 at http://www.opera.com
Powered by Outblaze
From finlayson at live555.com Thu Jul 27 12:57:49 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 27 Jul 2006 12:57:49 -0700
Subject: [Live-devel] Stream discovery
In-Reply-To: <1154015719.44c8e1e79effa@webmail.soton.ac.uk>
References: <1154015719.44c8e1e79effa@webmail.soton.ac.uk>
Message-ID:
>Is
>there a standard way of getting the server to list all the available
>transport streams?
No - unfortunately there's nothing like this in the RTSP standard.
The most 'standard' way to do something like this would be to use
HTTP - i.e., using a web server that delivers a HTML (or XML) page
that lists the available streams (e.g., as "rtsp://" URLs).
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From kenneth.tu at ebell.com.tw Thu Jul 27 18:04:36 2006
From: kenneth.tu at ebell.com.tw (Kenneth Tu)
Date: Fri, 28 Jul 2006 09:04:36 +0800
Subject: [Live-devel] Bad Network Situation
Message-ID: <001201c6b1e1$cbfad850$6401a8c0@kenneth>
Dear all
I have met a problem.
When I try to stream video by Streaming Server on my dsl network, It's data almost overload the bandwidth and made a serious packet loss.
I think this caused by my pool upload bandwidth.
Could I control the sending frequency by somewhat?? (Like RTCP)
Thanks in advance!!
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From finlayson at live555.com Thu Jul 27 18:23:08 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Thu, 27 Jul 2006 18:23:08 -0700
Subject: [Live-devel] Bad Network Situation
In-Reply-To: <001201c6b1e1$cbfad850$6401a8c0@kenneth>
References: <001201c6b1e1$cbfad850$6401a8c0@kenneth>
Message-ID:
>Dear all
>
>I have met a problem.
>
>When I try to stream video by Streaming Server on my dsl network,
>It's data almost overload the bandwidth and made a serious packet
>loss.
If you are trying to stream a video whose bitrate exceeds the
capacity of your network, then there's not much you can do - other
than reducing the bitrate of your video (i.e., reencoding it).
If you are streaming MPEG-1 or 2 video, then you can reduce the
bitrate by streaming I-frames only (search for "iFramesOnly" in the
code).
>
>I think this caused by my pool upload bandwidth.
>
>Could I control the sending frequency by somewhat?? (Like RTCP)
No - RTCP has nothing to do with this. Your video's bitrate is just
too high for your network.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From p.pradeep09 at gmail.com Fri Jul 28 02:21:06 2006
From: p.pradeep09 at gmail.com (Pradeep Kumar PALAPARTHY)
Date: Fri, 28 Jul 2006 14:51:06 +0530
Subject: [Live-devel] Receiving some packets after the session is complete
Message-ID: <8daef9c70607280221h1c677ae7q572cdeec2586193c@mail.gmail.com>
Hi,
Im using live 555 testprogs for streaming. I have started streaming using
testOnDemandRTSPServer.
I have developed some customized clients which uses mpeg4ip rtsp libraries.
When i connect to server live555 server, i found that after sending all the
rtp packets, still im receiving RTCP packets. I found this by running
ethereal and it is sending to data odd port numbers.
for example if my client ports are 1024-1025, im receiving packets to port
1025. I see same the thing for both Audio & Video.
In client logic, i waiting on time out since i dont have the playtime(npt
range). In live555 when i stream TS file(tranport streams) using rtsp i
found that the npt range is npt: 0 --
I gave a timeout of 5secs, but since im receiving the rtcp packets its not
getting timedout.
Can anyone help i this regard.
Thanks in advance,
Pradeep
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From dirkx at theveniceproject.com Fri Jul 28 04:22:07 2006
From: dirkx at theveniceproject.com (Dirk-Willem van Gulik)
Date: Fri, 28 Jul 2006 13:22:07 +0200
Subject: [Live-devel] Few more job openings around video codecs.
In-Reply-To:
References:
Message-ID: <44C9F35F.8020208@theveniceproject.com>
We've now got a few more job openings in Leiden, near Amsterdam, for
people who understand VLC, H.264, encoding and decoding, who have a
strong programming background, who absolutely get file formats, open
source, extracting headers, graphics munging and transcoding (or some
part thereof).
Experience with batch processing, XML, unix, scripting, large storage
systems or workflows is nice.
Full time, open ended Dutch employment contract, work permit generally
not an issue, can start this week, English language.
Thanks,
Dw.
--
Dirk-Willem van Gulik
From junker at rbg.informatik.tu-darmstadt.de Fri Jul 28 05:39:28 2006
From: junker at rbg.informatik.tu-darmstadt.de (Bertram Junker)
Date: Fri, 28 Jul 2006 14:39:28 +0200
Subject: [Live-devel] calling Live555 libraries from a C app under
Embedded Visual C++ 4.0
In-Reply-To: <49884.134.169.35.238.1153908524.squirrel@webmail.ibr.cs.tu-bs.de>
References: <1153494350.44c0ed4e45ade@webmail.soton.ac.uk> <55786.84.133.237.208.1153663529.squirrel@webmail.ibr.cs.tu-bs.de> <44C47125.3080106@rbg.informatik.tu-darmstadt.de>
<49884.134.169.35.238.1153908524.squirrel@webmail.ibr.cs.tu-bs.de>
Message-ID: <44CA0580.6030503@rbg.informatik.tu-darmstadt.de>
Hi Rodrigo,
my application is written in C++. I have renamed the main() function of
the Test-Application playSIP in playCommon.cpp from
main(int argc, char** argv)
to
mainplayCommon(int argc, char** argv)
and then in my own application I declare
extern int mainplayCommon(int, char**);
So I can call later in my application the complete testProgramm playSIP
like this:
char *progName = "playSIP";
const int argc = 12;
char *argOne = "-V";
char *argTwo = "-M";
char *argThree = "-A";
char *argFour = "8";
char *argFive = "-e";
char *argSix = "10";
char *argSeven = "-P";
char *argEight = "49211";
char *argNine = "-W";
char *argTen = "0";
char *argEleven = "172.16.1.100";
char * argv[argc];
argv[ 0] = progName;
argv[ 1] = argOne;
argv[ 2] = argTwo;
argv[ 3] = argThree;
argv[ 4] = argFour;
argv[ 5] = argFive;
argv[ 6] = argSix;
argv[ 7] = argSeven;
argv[ 8] = argEight;
argv[ 9] = argNine;
argv[10] = argTen;
argv[11] = argEleven;
mainplayCommon(argc,argv, this->m_frame);
Did this help you?
For an easier way to build my programm for Win32 and WinCE with one GUI
and one Code my programm is build-on the minimalSample from wxWidgets. I
don't know if this is realy intersting for your problem. But this sample
has one base code and you could build the same code with the same GUI
for Win32 and WinCE. In this sample if have include the code above.
Best regard,
Bertram
Rodrigo Weymar schrieb:
> Hi Bertram,
>
>
> is your application written in C or C++ ? Were you able to mix C and C++
> code and to declare C++ functions as extern "C" under EVC++ without using
> #ifdef __cplusplus ?
>
> If I try to use extern "C" under EVC++ without using #ifdef __cplusplus, I
> get a
>
> error C2059: syntax error : 'string'
>
> as explained in http://www.kbalertz.com/kb_133070.aspx
>
>
> regards,
> Rodrigo
>
>
>
>
>>Hi Rodrigo,
>>
>>i have written an SIP phone with RTP Stack from LIVE555 for Pocket PC /
>>Windows CE with EVC++ 4.0. Unfortunately I'm not able to solve your
>>problem but I can ensure you, that LIVE555 works fine under EVC++.
>>Further, I remember me fine, I have not set up any preprocessor directive.
>>
>>Best regards,
>>
>> Bertram
>>
>>
>>
>>>Hi all,
>>>
>>>probably this is not the right mailing list to ask, since my question is
>>>not directly concerning the Live555 libraries. But maybe someone can
>>>help
>>>me or give me an advice.
>>>
>>>I am trying to integrate the Live555 libs (I am interested in the RTSP
>>>support provided by these libs) into an C app, more exactly a streaming
>>>player for Pocket PC/WinCE420, which is written in C.
>>>
>>>I use Embedded Visual C++ 4.0 as IDE.
>>>
>>>I don't get any compilation errors, since from the C source code I am
>>>using the following preprocessor directive:
>>>
>>>#ifdef __cplusplus
>>>extern "C" {
>>>
>>>here I put the Live555 C++ classes
>>>
>>>}
>>>#endif
>>>
>>>
>>>What happens is that, when I run the app, the Live555 classes don't take
>>>any effect. That is, the Embedded Visual C++ C/C++ compiler seems to
>>>just
>>>ignore what is in between the preprocessor directives. It seems to have
>>>not linked the C++ classes to the C objs.
>>>
>>>The document in [1] says that the C++ runtime libraries should be
>>>explicitly
>>>linked to the app. The problem is that I can not figure out how Embedded
>>>Visual C++ manages that. I was not able to find any option in "Project
>>>Settings" or in "Tools -> Options" concerning that.
>>>
>>>I also googled for solutions, but was not able to find one.
>>>
>>>Would someone have previous experience with that? Could someone give me
>>>help, please ?
>>>
>>>
>>>Thanks a lot!
>>>
>>>Rodrigo
>>>
>>>
>>>[1] http://developers.sun.com/prodtech/cc/articles/mixing.html#linking
>>>
>>>_______________________________________________
>>>live-devel mailing list
>>>live-devel at lists.live555.com
>>>http://lists.live555.com/mailman/listinfo/live-devel
>>>
>>
>>
>>_______________________________________________
>>live-devel mailing list
>>live-devel at lists.live555.com
>>http://lists.live555.com/mailman/listinfo/live-devel
>>
>
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>
From frme at mail.ru Thu Jul 27 18:25:57 2006
From: frme at mail.ru (Frme)
Date: Fri, 28 Jul 2006 04:25:57 +0300
Subject: [Live-devel] Playing 3gp files.
Message-ID: <502794714.20060728042557@mail.ru>
Good afternoon!
I need to stream .3gp files on the test RTSP server, but as i noticed it
does not support this type of media files. What should I re-write or
what should i do in this case?
Thanks in advance.
From finlayson at live555.com Fri Jul 28 07:24:09 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 28 Jul 2006 07:24:09 -0700
Subject: [Live-devel] Few more job openings around video codecs.
In-Reply-To: <44C9F35F.8020208@theveniceproject.com>
References:
<44C9F35F.8020208@theveniceproject.com>
Message-ID:
This mailing list is for discussion of development using the "LIVE555
Streaming Media" libraries.
Job postings (unless they relate specifically to the LIVE555
software) are inappropriate for this mailing list. Please don't do
this again.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From larissalucena at gmail.com Fri Jul 28 07:29:00 2006
From: larissalucena at gmail.com (Larissa Lucena)
Date: Fri, 28 Jul 2006 11:29:00 -0300
Subject: [Live-devel] JPEG video
In-Reply-To: <440165240607270430lc299512o9645f56d41448a80@mail.gmail.com>
References: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
<440165240607270430lc299512o9645f56d41448a80@mail.gmail.com>
Message-ID: <440165240607280729w6700006dyd336ec6ee4e216fd@mail.gmail.com>
Hi there,
somebody has a code exaple of JPEGVideoSource' subclass that uses a video
file like source???
Thanks in advance.
Larissa
On 7/27/06, Larissa Lucena wrote:
>
> P/ transmitir via RTP...
>
>
>
>
> On 7/26/06, Loreno Oliveira wrote:
> >
> > Oq eh que tu precisa fazer? eh pra reproduzir ou transmitir via RTP?
> >
> >
> >
> > On 7/26/06, Larissa Lucena < larissalucena at gmail.com > wrote:
> >
> > > Hi there,
> >
> > I'm developing an application that I have to use videos like various
> > JPEGs. I'm using the live, but I don't know how to do it even so... could
> > you help me? Maybe an example code?
> >
> > Thanks,
> >
> > Larissa
> >
> >
> > --
> > "O maior prazer do inteligente ? bancar o idiota
> > diante de um idiota que banca o inteligente".
> >
> > _______________________________________________
> >
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> >
> >
> >
> >
> > _______________________________________________
> > live-devel mailing list
> > live-devel at lists.live555.com
> > http://lists.live555.com/mailman/listinfo/live-devel
> >
> >
> >
>
>
> --
> "O maior prazer do inteligente ? bancar o idiota
> diante de um idiota que banca o inteligente".
>
--
"O maior prazer do inteligente ? bancar o idiota
diante de um idiota que banca o inteligente".
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From finlayson at live555.com Fri Jul 28 07:35:13 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 28 Jul 2006 07:35:13 -0700
Subject: [Live-devel] Receiving some packets after the session is
complete
In-Reply-To: <8daef9c70607280221h1c677ae7q572cdeec2586193c@mail.gmail.com>
References: <8daef9c70607280221h1c677ae7q572cdeec2586193c@mail.gmail.com>
Message-ID:
>Hi,
>
>Im using live 555 testprogs for streaming. I have started streaming
>using testOnDemandRTSPServer.
>I have developed some customized clients which uses mpeg4ip rtsp
>libraries. When i connect to server live555 server, i found that
>after sending all the rtp packets, still im receiving RTCP packets.
See the code for "afterPlayingStreamState()" in
"liveMedia/OnDemandServerMediaSubsession.cpp". If a stream has a
known duration (which is usually the case if you're streaming from a
file), then the RTP session is kept alive even after the input source
ends - in case the client subsequently wants to seek backwards to an
earlier point in the stream.
Note, however, that a client-issued RTSP "BYE" command will always
close the RTP session.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From finlayson at live555.com Fri Jul 28 07:59:56 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Fri, 28 Jul 2006 07:59:56 -0700
Subject: [Live-devel] JPEG video
In-Reply-To: <440165240607280729w6700006dyd336ec6ee4e216fd@mail.gmail.com>
References: <440165240607260707g7bfe1cd7v20e04a134d206420@mail.gmail.com>
<440165240607270430lc299512o9645f56d41448a80@mail.gmail.com>
<440165240607280729w6700006dyd336ec6ee4e216fd@mail.gmail.com>
Message-ID:
>Hi there,
>
>somebody has a code exaple of JPEGVideoSource' subclass that uses a
>video file like source???
Yes - see the source code for the "ElphelStreamer" application (used
for streaming from an Elphel motion-JPEG video camera) at
See also
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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From dbikash at gmail.com Sun Jul 30 22:38:18 2006
From: dbikash at gmail.com (Deeptendu Bikash)
Date: Mon, 31 Jul 2006 11:08:18 +0530
Subject: [Live-devel] Streaming ADTS AAC in LATM format
Message-ID: <29758ec70607302238qf76bd71u200cc415ed5ce85@mail.gmail.com>
Hello,
Currently Live packetizes ADTS AAC as per RFC 3640 (MPEG-4 Generic). How can
I write code to packetize it as per RFC 3016 (LATM)?
Live has a LATM RTP Sink but no LATM File Source. Can you give some details
as to how to do it? Specially the config part?
Thanks,
Deeptendu
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From finlayson at live555.com Mon Jul 31 01:19:41 2006
From: finlayson at live555.com (Ross Finlayson)
Date: Mon, 31 Jul 2006 01:19:41 -0700
Subject: [Live-devel] Streaming ADTS AAC in LATM format
In-Reply-To: <29758ec70607302238qf76bd71u200cc415ed5ce85@mail.gmail.com>
References: <29758ec70607302238qf76bd71u200cc415ed5ce85@mail.gmail.com>
Message-ID:
>Hello,
>
>Currently Live packetizes ADTS AAC as per RFC 3640 (MPEG-4 Generic).
>How can I write code to packetize it as per RFC 3016 (LATM)?
You would need to:
1/ Write a filter that converts raw AAC audio frames into
LATM-formatted AAC audio frames (i.e., by turning it into an
"AudioMuxElement"), and
2/ Use "MPEG4LATMAudioRTPSink" (instead of "MPEG4GenericRTPSink") as
the RTP sink
So, the data chain would become:
ADTSAudioFileSource -> yourRawToLATMFormatFilter -> MPEG4LATMAudioRTPSink
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
From tooluck at tom.com Mon Jul 31 01:39:21 2006
From: tooluck at tom.com (tooluck at tom.com)
Date: Mon, 31 Jul 2006 16:39:21 +0800
Subject: [Live-devel] about the openrtsp
Message-ID: <002101c6b47c$d1f5a940$5801a8c0@luoqt>
hello,I used the openrtsp to get record from a server,and I get the file 1.mp4 and 1.avi,but the file could not be streaming by the helix server(apply corp.),
I think it is because there is no index in the file 1.mp4 or 1.avi.
how can I do now! thanks!
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From frme at mail.ru Sun Jul 30 16:04:59 2006
From: frme at mail.ru (Frme)
Date: Mon, 31 Jul 2006 02:04:59 +0300
Subject: [Live-devel] Stream .3gp files
Message-ID: <1116252944.20060731020459@mail.ru>
Good afternoon,
I asked already on this topic but maybe my question wa not very clear:( .
so, I need to extend sample RTSPserver to have ability from my RTSP
client to reach some .3gp content, which should be sreamed by this server.
What is the proper way to achieve this? or live-555 library does not
supports .3gp content at all? :(
Euegene.