> On Jan 31, 2024, at 5:08âŻAM, Denis Gottardello
> wrote:
>
>
> Hi, I have this output from the server. What it meas?
>
> MultiFramedRTPSink::afterGettingFrame1(): The input frame data was too large
> for our buffer size (100176). 38444 bytes of trailing data was dropped!
> Correct this
Hi, I have this output from the server. What it meas?
etBuffer::maxSize" to at least 179091, *before* creating this 'RTPSink'.
(Current value is
10.)
MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds
the
client's buffer size (10). 38444 bytes of trailing
> On Jan 10, 2023, at 2:08 AM, g.jaegy wrote:
>
> Why that ? I understand it might require a higher buffer maybe since packets
> might be reordered/resent/etc., any other reason ?
Streaming over TCP is less data efficient, and can lead to much higher latency
than regular UDP streaming. See
ject: Re: [Live-devel] question RTSP server
> On Jan 9, 2023, at 11:53 AM, Ross Finlayson wrote:
>
> Otherwise, if your client can reach the server via the normal RTSP ports (554
> or 8554), then you just use RTSP as usual. By default, this will give you
> RTP-over-UDP.
ayson
Sent: Monday, January 9, 2023 8:54 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] question RTSP server
> On Jan 9, 2023, at 11:41 AM, g.jaegy wrote:
>
> Perfect, that what I've done and it's working like a charm đ I've indeed
&g
> On Jan 9, 2023, at 11:53 AM, Ross Finlayson wrote:
>
> Otherwise, if your client can reach the server via the normal RTSP ports (554
> or 8554), then you just use RTSP as usual. By default, this will give you
> RTP-over-UDP. But if your have a firewall that blocks UDP packets, your
> cli
> On Jan 9, 2023, at 11:41 AM, g.jaegy wrote:
>
> Perfect, that what I've done and it's working like a charm đ I've indeed
> extracted an underlying class with one instance shared across all my custom
> source inputs, so that input sources can be destroyed/created at anytime.
> Works great.
uary 9, 2023 6:24:49 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] question RTSP server
> On Jan 9, 2023, at 6:21 AM, g.jaegy wrote:
>
> Fantastic, thanks a lot for that quick and very precise answer.
>
> I've managed to update the cod
> On Jan 9, 2023, at 6:21 AM, g.jaegy wrote:
>
> Fantastic, thanks a lot for that quick and very precise answer.
>
> I've managed to update the code quickly, however, I'm just facing a little
> issue : I am not completely sure what the "reuseFirstSource" parameter in the
> OnDemandServerMedi
tanciate a new frame, but use the same single input
source object ?
Thanks a lot !
-Original Message-
From: live-devel On Behalf Of Ross Finlayson
Sent: Monday, January 9, 2023 12:27 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] question RTSP serve
> On Jan 9, 2023, at 2:57 AM, g.jaegy wrote:
>
> Hi,
> I have a RTSP server that can stream a H265 stream from a custom H265 packet
> source, used by a âH265VideoStreamDiscreteFramerâ object (basically a live
> screen capture of a Vulkan application sent down to NVenc which generated the
>
Hi,
I have a RTSP server that can stream a H265 stream from a custom H265 packet
source, used by a "H265VideoStreamDiscreteFramer" object (basically a live
screen capture of a Vulkan application sent down to NVenc which generated the
H265 packets). It uses a "PassiveServerMediaSubsession" objec
Sorry, but our software is C++ only, so we canât help you with a RTSP client
written in Python. We do, however, have two RTSP client applications - written
in C++ - that you can use: âtestRTSPClientâ, and âopenRTSPâ (see
http://www.live555.com/openRTSP )
Ross Finlayson
Live Networks, Inc.
htt
Hello! I encountered a problem when using a python program to get the rtsp
video stream I built by myself.I successfully built the live555 media server on
the Linux server and put the stream file in the /mediaServer/ directory.I want
to use cap = cv2.VideoCapture("rtsp://IP:8554/data.264") to ge
> On Jan 12, 2021, at 7:39 PM, Ross Finlayson wrote:
>
> Because your use case seems reasonable, I think the best solution is for us
> to update the code so that âlookupServerMediaSession()â is an asynchronous
> operation (taking a âcallbackâ parameter that will get called when the the
> loo
> And now I still have a possible option: implementing a event-pool
> "TaskScheduler". But I worry whether this is feasible, since the whole
> library is designed basing on event-driven with one thread.
No, this wonât work; LIVE555 application code must be single-threaded.
Because your use case
Note that "GenericMediaServer::lookupServerMediaSessionâ is part of the library
code, and thus is not meant to be modified. Instead, you should subclass
âRTSPServerâ, and reimplement the virtual function âlookupServerMediaSessionâ in
your subclass. (This is what we did for the âDynamicRTSPSer
> On Jan 11, 2021, at 5:56 PM, Mit Shan wrote:
>
> I want to achieve the following function in live555 RTSP server:
>
> (a). Block for some time in the method
> "GenericMediaServer::lookupServerMediaSession"
Note that "GenericMediaServer::lookupServerMediaSessionâ is part of the library
cod
I want to achieve the following function in live555 RTSP server:
(a). Block for some time in the method
"GenericMediaServer::lookupServerMediaSession", when processing the
"DESCRIBE" request (and before we response to it), while having live555
RTSP server running normally with other streams da
--- Begin Message ---
Thanks for your response. I will provide more context here.
I followed the code in the testRTSPClient.cpp, and defines our own client as
well as data sink. The current setup is I have a local device running ubuntu
18.04 and an ip camera(have two streams), both of them conn
> On Dec 11, 2020, at 9:14 AM, Zhang, David via live-devel
> wrote:
>
> Recently I have met a problem when connecting multiple RTSP streams in our
> application. When I connect 9 streams, the application will work properly.
> But when I increase the number of streams to 10 or more, the strea
--- Begin Message ---
Hi, I am David Zhang from Amazon.
Recently I have met a problem when connecting multiple RTSP streams in our
application. When I connect 9 streams, the application will work properly. But
when I increase the number of streams to 10 or more, the streams will first run
for a
Youâre going to have to be a lot more specific about why the â-mâ option is not
working for you.
(And Remember, You Have Complete Source Code.)
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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At this time, what I'm trying to do is create a snapshot of an openRTSP stream
(just capture a SINGLE frame(NOT CONTINOUSLY) for post processing for facial
recognition.
I've tried this option:
Outputting each frame into a separate file
If the "-m" option is given, each incoming 'frame' will be wr
Ah ok, so this is a feature in the new 2016.11.17. I was using
2016.11.06 before and wondered why it didn't work anymore.
We use qmake as build system so I'm using a custom .pro file to build
livemedia.so together with our other code.
Op 17/11/2016 om 21:14 schreef Ross Finlayson:
Frederik,
> Frederik, there was a new .cpp file (IIRC it was RTSPServerRegister.cpp in
> the liveMedia dir) in one of the recent live555 releases, Ross split the
> register implementation across multiple files.
Yes. You need to run âgenMakefilesâ again to generate a new Makefile that
mentions this new f
Frederik, there was a new .cpp file (IIRC it was RTSPServerRegister.cpp in the
liveMedia dir) in one of the recent live555 releases, Ross split the register
implementation across multiple files.
HTH,
Ralf
>>> Frederik De Ruyck 17/11/16 7:03 PM >>>
Ross,
in what version did you drop the follow
Ross,
in what version did you drop the following?
void RTSPServer::implementCmd_REGISTER(char const* /*url*/, char const*
/*urlSuffix*/, int /*socketToRemoteServer*/,
Boolean /*deliverViaTCP*/, char const*
/*proxyURLSuffix*/) {
// By default, this function is a 'noop'
You may find it useful to use the âDeviceSourceâ code as a model; see
âliveMedia/DeviceSource.cppâ.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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Hello,
I have problem with creating server said in thread. What I did:
Class A that initiate everything + start event loop
Class B which inherits from OnDemandServerMediaSubsession with redefined:
createNewRTPSink and CreateNewStreamSource which return my C class
Class C which inherits from Audi
> I'm looking for the function that makes the call to control play with a
> definite time
> eg play from one minute
Your question wasn't clear, but I presume that you're asking about a *RTSP
client* function. (Remember that the code also implements a RTSP server, and a
RTSP proxy.)
The func
hello,
I'm looking for the function that makes the call to control play with a
definite time
eg play from one minute
cordially,
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> I googled and I found the open source site from that camera (1), I found a
> sheet containig live streaming information (2). There I found Live555.
[...]
> (1): http://gopro.com/support/open-source
> (2):
> http://wpcdn.gopro.com.s3.amazonaws.com/wp-content/uploads/2013/01/live.2012.02.04.tar.g
Hello!
I am a GoPro camera user, and I am a bit dissapointed because of the lag
produced in the WiFi Live Streaming.
The GoPro streams to a server (10.5.5.9:8080) and the preview is here:
http://10.5.5.9:8080/live/amba.m3u8
I googled and I found the open source site from that camera (1), I found
Okay, I did some more testing of the problem.
Step one: I set the max_key_interval in the encoder to 25 (one per second
and current fps). Xvid sends a Bitstream header with each keyframe, so the
streamer shouldn't have to wait for more than a second max for that
information.
Step two: I wrote a
Chris,
I've gotten a very similar pipe thing working. Ross was invaluable in
helping. Here are two points you may need to recall, to finally get it
working properly.
This advice relates to using H.264. It may only partially relate to your
application.
1) You must make sure that you're sending
> As far as Windows developers who use gmail...I can understand the
> anti-windows sentiment, but I don't know what you have against gmail...
This is explained clearly in the FAQ (that everyone was asked to read before
posting to the mailing list :-)
> And I code in Windows because I'm paid to
Yes, the encoder is another process (which is in turn reading raw frames
from yet another process). I'll test my pipe code in the encoder; I'm
doing it differently from in the 3D program going to the encoder.
As far as Windows developers who use gmail...I can understand the
anti-windows sentiment
> My question is this: can I safely undefine READ_FROM_FILES_SYNCHRONOUSLY
NO! You should not modify the supplied source code. (Windows developers who
use "@gmail.com" email addresses should especially not modify the supplied
source code :-)
The whole point of this code is that, in Windows, r
> I implemented the DummySink, but I think I need to modified the data I'm
> getting before passing it to FFMPEG
No you don't - not for MPEG Transport Stream data.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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Thanks for the quick response. One other question:
I implemented the DummySink, but I think I need to modified the data I'm
getting before passing it to FFMPEG (header file and other stuff. I've read
that's necessary for H.264, but not sure for MPEG-TS). Is that right? If is the
case, would you
use
Subject: Re: [Live-devel] Question About PAT/PMT in MPEG2
I'm building and iOS application, where I'm using a RTSP Client (based on
testRTSPClient) to receive a MPEG2 stream
Because you refer to "PAT" and "PMT" information, I assume you're referring to
a MPEG-
> I'm building and iOS application, where I'm using a RTSP Client (based on
> testRTSPClient) to receive a MPEG2 stream
Because you refer to "PAT" and "PMT" information, I assume you're referring to
a MPEG-2 *Transport* stream. (There are other kinds of MPEG-2 streams as well.)
> and then I'm
Hi,
I'm building and iOS application, where I'm using a RTSP Client (based on
testRTSPClient) to receive a MPEG2 stream and then I'm trying to decode the
frames with help of FFMPEG. My question is: Is it possible to obtain the PAT
and PMT information before I pass the frames to FFMPEG with he
Hello sir,
i made an streamer using your libraries which streams the live video from
camera
int initLm555Settings(void)
{
scheduler = BasicTaskScheduler::createNew();
env = BasicUsageEnvironment::createNew(*scheduler);
destinationAddressStr
#ifdef USE_SSM
= "232.255.42.42";
#else
= Streaming
On 12/28/2012 14:04, feriel ben ghorbel wrote:
Actually and as you said Warren I used Embedding VLC plugin in a
HTML Page(browser: Mozilla Firefox) using Apache2 and all is
included with Ubuntu(11.10)
my problem is when I reach the Html Page ; the stream ".ts" is not
working and usually M
> Ross,
> Can Live555 Media Server serv media over http ?
No (except for "RTSP-over-HTTP", which is not what you're talking about here).
Therefore, this thread has become off-topic for this mailing list.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
_
- Original Message -
From: "feriel ben ghorbel"
To: "LIVE555 Streaming Media - development & use"
Sent: Friday, December 28, 2012 10:04 PM
Subject: Re: [Live-devel] Question live555MediaServer
Hi
At first Thanks Warren Young and Tayeb Meftah for your quick answers.
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Warren Young
Sent: Thursday, December 27, 2012 6:30 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Question live555MediaServer
On 12/26/2012 07:38, feriel ben ghorbel wrote:
> Bonjour ,
You're going to g
---
From: "feriel ben ghorbel"
To:
Sent: Wednesday, December 26, 2012 3:38 PM
Subject: [Live-devel] Question live555MediaServer
Bonjour ,
Est ce que en utilisant "live555MediaServer" peut on afficher un flux
ou plusieurs flux ".ts" Ă travers un browser ??
Cordia
On 12/26/2012 07:38, feriel ben ghorbel wrote:
Bonjour ,
You're going to get more help if you use English than French.
Est ce que en utilisant "live555MediaServer" peut on afficher un flux
ou plusieurs flux ".ts" Ă travers un browser ??
VLC has a browser plugin that can catch MPEG TS stream
Hi.
I have a question about RTSP PAUSE / RESUME timestamp.
I'm using live555(ADTSAudioFileServerMediaSubsession.cpp &
H264VideoStreamDiscreteFramer.cpp)
If One video frame is truncated by PAUSE / RESUME request, how to use
timestamp?
RTP packet of a same frame's timestamps are sa
Hi to all...
I'm trying to modify the openRTSP.
I need to save the the rtsp stream into files with one minute length.
For do this I need modify the QuickTimeFileSink class, adding the
changeoutputfile method that I have to call each minute...
Have you some ideas about how I can do this ?
M
that
makes sense.
From: live-devel-boun...@ns.live555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Monday, August 27, 2012 3:24 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] Question about live streaming
It does not. I am g
> It does not. I am going to be streaming globally and am using a global server
> to direct all the traffic, creating rtsp streams for each device.
I'm not sure I understand this, but from your description, I suspect that you
don't want to be using RTSP at all. RTSP is used for one-way streamin
It does not. I am going to be streaming globally and am using a global server
to direct all the traffic, creating rtsp streams for each device.
Original message
Subject: Re: [Live-devel] Question about live streaming
From: Ross Finlayson
To: LIVE555 Streaming Media
> I am currently building an Android application designed to stream encoded
> video (live) from each phone to a different phone, and I play to use Live555
> as the server. I read that to support live streaming to edit the
> OnDemandRTSPServer, as well as make a few other changes. I did said chan
Hello,
I am currently building an Android application designed to stream encoded
video (live) from each phone to a different phone, and I play to use Live555
as the server. I read that to support live streaming to edit the
OnDemandRTSPServer, as well as make a few other changes. I did said chan
Hi.
I'm using LIVE555 Proxyserver.
Can I multiplexing RTSP/RTP stream to mpeg2-ts using
MPEG2TransportStreamMultiplexor class in LIVE555?
The stream is received from back-end LIVE555 stream server.
(video stream: H.264, audio stream: AAC)
Thanks for your interest.
> Forward / Backward random seeking is supported in LIVE555 Proxy server?
No, it's not - because the purpose of the "LIVE555 Proxy Server" is to share a
single 'back-end' stream amongst possibly several concurrent 'front-end'
streams.
If you want to seek within a RTSP stream, then don't proxy i
Hi.
Forward / Backward random seeking is supported in LIVE555 Proxy server?
In my case,,
Seeking bar is activated by modifying the "virtual float duration( ) const;
however, the command of client( "PAUSE" or "PLAY") did not send to back-end
LIVE555 Server.
To do this, what shall I mod
Hello Ross
Sorry for the late reply and thanks alot for your answer
I have tried unicast
server said:
"h264ESVideoTest" stream, from the file "test.264"
Play this stream using the URL "rtsp://192.168.1.4:8554/h264ESVideoTest"
I do have a test.264 file in that folder
Client said:
*2012-02-08 21
Also, your client is using a very old version of the "LIVE555 Streaming Media"
software. You should upgrade it if you can.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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> Received SETUP response: RTSP/1.0 461 Unsupported Transport
This happens because the server's stream is multicast, but your client is
asking for it to be streamed via RTP-over-TCP, which you can't do for multicast
streams.
If your client *does not* request RTP-over-TCP streaming (e.g., if you
Forgot to mention that I m using testH264VideoStreamer.
On 3 February 2012 14:37, James Zhang wrote:
> Hello Ross
>
> I m using live555 test server to streaming a h264 video form hard disc
>
> from terminal it looks like
>
> Play this stream using the URL "rtsp://172.28.31.103:8554/testStream"
>
Hello Ross
I m using live555 test server to streaming a h264 video form hard disc
from terminal it looks like
Play this stream using the URL "rtsp://172.28.31.103:8554/testStream"
Beginning streaming...
Beginning to read from file...
My client get:
**
*2012-02-03 14:32:37.376 rtsp[51110:6c03]
> I got a question. Lots of documents mentioned that (7) sps, (8) pps, (6) sei.
> what is that mean?
They're defined in the H.264 specification: ISO/IEC 14496-10, I think.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
live-devel mai
Hello everyone
I got a question. Lots of documents mentioned that (7) sps, (8) pps, (6)
sei. what is that mean?
Nal unit is a binary set, 7 means position 7 or 0111 or something else?
Thanks alot
Best regards
James
On 31 January 2012 23:17, James Zhang wrote:
> Hello everyone
>
> Thanks for
Hello everyone
Thanks for everybody's suggestion.
Looks i have made it start to do something instead on no frame, fail to
decode. I m keep getting something like this. Is it because of i m putting
the wrong NAL units to decoder?
My nal structure is
0x1sps0x1pps
*[h264 @ 0x700f400]slice type to
>
> >
> > SPropRecord * data to a NSData and send into extradata to decode?
>
> I don't know what a "NSData" is (it's apparently something outside our
> libraries), but I hope it should be obvious from the implementation of the
> function in "liveMedia/H264VideoRTPSource.cpp" how it works.
>
>
"NS
James,
You can parse sprop parameters in order to get meaningfull stream
info [ stream width, height etc] if you need.
You can get details from here:
http://stackoverflow.com/questions/6394874/fetching-the-dimensions-of-a-h264video-stream
Best Wishes
Novalis
2012/1/31 Ross Finlayson :
> After
> After parseSPropParameterSets, I should be able to get sps and pps(in binary
> format?)
Yes, and that's exactly what the "sPropBytes" field is! It's a pointer to an
array of "sPropLength" bytes (i.e., binary).
>
> But when i use print command to print the strings, I got some wired data.
No
Hello Ross
Thank you very much for your fast and nice reply.
After parseSPropParameterSets, I should be able to get sps and pps(in
binary format?)
But when i use print command to print the strings, I got some wired data. I
m not sure why it looks like that:
code:
for(unsigned i=0; i
> I have
G'day James, it's nice to hear from another University of Auckland person.
> I have a question about parseSPropParameterSets() function.
>
> Based on my understanding, I think this function will read in the SPS and PPS
> data,
Yes, it will read in a coded ASCII string that represents the SPS a
Hello everyone
I have a question about parseSPropParameterSets() function.
Based on my understanding, I think this function will read in the SPS and
PPS data, then do base64 decode to generate a nal unit.
I have fit the SPS and PPS data by using this
unsigned int num=0;
SPropRecord *
sps=pars
> What I mean by âtaggingâ is essentially the ability to bookmark a place in
> the video with something (maybe a timestamp) and then the ability to fast
> forward or reverse to that point in the video.
Well, our FAQ has an entry that talks about our support for 'trick play'
(seeking, fast-forwa
ive555.com
[mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson
Sent: Thursday, December 22, 2011 9:53 PM
To: LIVE555 Streaming Media - development & use
Subject: Re: [Live-devel] question
Does this code allow tagging of the incoming video data?
Can you explain more wh
> Does this code allow tagging of the incoming video data?
>
Can you explain more what you mean by "tagging"?
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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Does this code allow tagging of the incoming video data?
Beth Turk
DRS C3A Engineering
246 Airport Road
Johnstown, PA 15904
beth.t...@drs-c3.com
Office: 814-534-8705
Cell:814-242-4943
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> I've used the example "testH264VideoToTransportStream" as a basis for
> generating a H264-TS-over-RTP stream of packets.
> I have used Wireshark to view the stream of RTP packets, and they appear to
> be correctly filled with an integer number of 188 byte Transport-Stream
> packets.
>
> I am
I've used the example "testH264VideoToTransportStream" as a basis for
generating a H264-TS-over-RTP stream of packets.
I have used Wireshark to view the stream of RTP packets, and they appear to be
correctly filled with an integer number of 188 byte Transport-Stream packets.
I am trying to fur
> However there's something that I'm not sure with. When a wav file or raw pcm
> audio data is streamed, I found that a buffer will be used to preload some
> data (few seconds, related with audio file's bitrate) before MPlayer starts
> playing.
This is a MPlayer issue; there is no such bufferin
Hello,
I've tried to use live media libraries with MPlayer for streaming audio
files over RTSP and it is successful.
However there's something that I'm not sure with. When a wav file or raw
pcm audio data is streamed, I found that a buffer will be used to preload
some data (few seconds, related w
> on client side we have hasBeenSynchronizedUsingRTCP to determine RTCP sync
> event
> what about server side, i also need determine this event, for proper data
> generation in my framer clas.
Sorry, but your question makes no sense. The reception of the first RTCP "SR"
packet by a receiver is
on client side we have * hasBeenSynchronizedUsingRTCP to determine RTCP sync
event*
what about server side, i also need determine this event, for proper data
generation in my framer clas.
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I am testing RTSP/RTP with liveMediaserver with MPEG2-TS.
I made index file and the file for trickmode using
testMPEG2TransportStreamTrickPlay.exe
For example, original contents is avatar.ts
index file is avatar.tsx and file for trickplay is avatar2.ts
You have misunderstood the purpose of th
Hello all,
I am testing RTSP/RTP with liveMediaserver with MPEG2-TS.
I made index file and the file for trickmode using
testMPEG2TransportStreamTrickPlay.exe
For example, original contents is avatar.ts
index file is avatar.tsx and file for trickplay is avatar2.ts
I am using the VLC and my player
Ross,
Actually I did not modified the source code. I created subclasses of
specified objects and created an OnDemandContinuousMediaSubsession object.
Regards,
Jonathan
On 12/23/2010 02:32 PM, Ross Finlayson wrote:
Because you have modified the supplied source code, you can expect no
further
Because you have modified the supplied source code, you can expect no
further help from me. Sorry.
--
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
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Ross,
I finally make it working. I have a weird behavior with VLC, but I'm
trying to fix it.
I did it adding to the afterPlayingStreamState function (in
OnDemandServerMediaSubsession), a call to seekStream, to the position 0
and a call to RTPSink->startPlaying().
The problem that I had wit
Now the problem is where to make the stream to loop to the start of
file... you said:
"You could do this, but you'd need to write a new "FramedFilter"
subclass that sits
in front of your "ByteStreamFileSource" class (and presents the
illusion of delivering a single, unbroken stream to the down
De: Ross Finlayson
Fecha: Viernes, Diciembre 10, 2010 5:30 am
Asunto: Re: [Live-devel] Question about trick play, server side
> >I just found a "bit" change... I just realized that the live555
> >version I can use here is 2009.02.13.
>
> Sorry, but support is given o
I just found a "bit" change... I just realized that the live555
version I can use here is 2009.02.13.
Sorry, but support is given only for the latest version of the code
only. (The version that you're using has many, many bugs that were
fixed in subsequent versions.)
PS: I'm not able to up
n
- Mensaje original -
De: Jonathan Granade
Fecha: Miércoles, Diciembre 8, 2010 8:19 pm
Asunto: Re: [Live-devel] Question about trick play, server side
> Ross,
>
> Sorry by being late in reply but I was pretty busy.
>
> Thanks for your reply, it sounds very very good and I'
Ross,
Sorry by being late in reply but I was pretty busy.
Thanks for your reply, it sounds very very good and I'm starting to test it.
About stream N minutes of a file, I dont want to "cut" the original
video, I think that I can modify the headers or something to make the
client think that th
I checked the code mentioned, and it works as expected.
But now I have a question... is possible to do the same (stream only
N mins of file, or make an infinite loop) from the server side?
Well, if you want to only stream N minutes of a file - without the
client asking you to do this - then th
1:32 am
Asunto: Re: [Live-devel] Question about trick play, server side
> >I'm new at live555, i read doxygen references and make some tests
> to
> >start understanding how it works.
> >Now i plan something to do, i think simple, I want to stream a
> video
>
I'm new at live555, i read doxygen references and make some tests to
start understanding how it works.
Now i plan something to do, i think simple, I want to stream a video
file (.mpg) on demand, but with one feature... I want to specify the
duration to stream in seconds, or make it an infinite l
Hello,
I'm new at live555, i read doxygen references and make some tests to start
understanding how it works.
Now i plan something to do, i think simple, I want to stream a video file
(.mpg) on demand, but with one feature... I want to specify the duration to
stream in seconds, or make it an in
I view adding a morphological filter to my network streaming library
as a rather substantial kludge, so I'll chalk this up to "difference
of opinion."
Well maybe. However, because I was the person who designed the
library and its architecture, some opinions are more valuable than
others :-)
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