[Live-devel] Please unsubscribe me from all mailing lists!

Hi! Please unsubscribe me from all mailing lists! My email: ad...@soft-collection.com Thank you! ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

[Live-devel] I-frame or P-frame

Hi! Thank you very much for your help with SPS and PPS. Please, can you suggest me where can I find information about frame type. Whether it is an I-frame or P-frame. Thank you! ___ live-devel mailing list live-devel@lists.live555.com http:/

[Live-devel] Please suggest me how can I organize decoding of the stream.

Hi! Please suggest me how can I organize decoding of the stream. Now I can get the h264 frames from RTSP as byte arrays, their sizes in bytes, mediumNames and codecNames of these frames. 1. But which test*.cpp file have I use as a base in order to decode these frames? Is it openRTSP.cpp or

[Live-devel] Please help me to understand how I can retrieve the Frame Width, Frame Height and Frame Type (I or P frame).

Hello! Thank you very much for this project! Please help me to understand how I can retrieve the Frame Width, Frame Height and Frame Type (I or P frame). My code example is based on testRTSPClient.cpp I am in the function DummySink::afterGettingFrame fSubsession.videoWidth() and fSubsession.vi

Re: [Live-devel] Wrong reconstructed JPEG Header after RTP Streaming

adjusting the qFactor value. I didn't get perfect result though. So now I have to figure out how I can compute the qFactor from my Encoder's "quality Factor" ... qFactor is between 0 (worst) and 100 (best)? Ursprüngliche Nachricht ----Von: "Bommes, Michae

[Live-devel] Wrong reconstructed JPEG Header after RTP Streaming

D5 D6 D7 D8 D9 DA E2 E3 E4 E5 E6 E7 E8 E9 EA F2 F3 F4 F5 F6 F7 F8 F9 FA FF DA 00 0C 03 01 00 02 11 03 11 00 3F 00 Is a wrong qFactor value the reason for the wrong header, or how/where do I have to change my JpegSource class or the server / session to stream to f

[Live-devel] Can't receive packets from live555 server on ARM board (testH264VideoStreamer binary)

Thank you Ross Finlayson, I changed destinationAddress.s_addr = chooseRandomIPv4SSMAddress(*env); to destinationAddress.s_addr = our_inet_addr("192.168.31.1"); for testing and now it works, so missing multicast support really is the problem here. What is the right way to go on allowing both: m

[Live-devel] Can't receive packets from live555 server on ARM board (testH264VideoStreamer binary)

Streamer sample server settings) Thanks in advance Michael ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] Creating live tsx file or Updating existing TSX file

, having VLC player (merely as an example) connect to our Media Server and perform the near-real time trick play. I hope that makes sense and better clarifies our goal. That is why we are looking into the tsx file generation possibilities. Thank you, Michael Chapman From: live-devel [mailto:live

Re: [Live-devel] EXTERNAL: Re: Creating live tsx file or Updating existing TSX file

without the requested feature. Thank you, Michael Chapman From: live-devel [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Sunday, November 16, 2014 1:07 PM To: LIVE555 Streaming Media - development & use Subject: EXTERNAL: Re: [Live-devel] Creating live tsx fil

[Live-devel] Creating live tsx file or Updating existing TSX file

dexer via >subclassing (assuming the tsx file structure and MediaServer logic supports >this case)? Thank you, Michael Chapman ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

[Live-devel] Problem setting rtpmap on Flir A320 thermal camera.

sampling=mono; width=160; height=120; depth=16 a=rtpmap:102 raw/9 a=framesize:102 320-240 a=fmtp:102 sampling=mono; width=320; height=240; depth=16 a=rtpmap:103 raw/9 a=framesize:103 160-120 a=fmtp:103 sampling=mono; width=160; height=120; depth=16 Thanks in advance Best regards Michael

Re: [Live-devel] Changes to make it easier to subclass ServerMediaSession and set packet buffer size

vel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: 20 February 2014 17:39 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Changes to make it easier to subclass ServerMediaSession and set packet buffer size Michael, To begin with, it's important to note

[Live-devel] Changes to make it easier to subclass ServerMediaSession and set packet buffer size

Hi, I would like to request a few changes to liveMedia to assist with subclassing and server instantiation. Some background: I'm working on a network camera that needs to be ONVIF profile G compliant. It records matroska video clips which are then represented to an ONVIF client as a single "Re

Re: [Live-devel] Media level SDP lines exceed memory allocation

Thanks Ross. Tested and working in 2014.02.10. Mike From: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: 10 February 2014 11:51 To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Media level SDP lines exceed me

[Live-devel] Media level SDP lines exceed memory allocation

Hi, I'm a new subscriber working on security cameras. I'm experiencing an RTSP server crash in the DESCRIBE phase which I have traced to the function ServerMediaSession::generateSDPDescription(). In this function, I think the sdpLength is being calculated based on the length of the session-leve

Re: [Live-devel] getNormalPlayTime returning negative value

I was performing the timestamping using gettimeofday as the timestamps, but before temporarily queuing the samples. Once I moved the timestamping to the track's deliverFrame method the problem was resolved. Thanks for your help. (First, when replying to a mailing list digest, please use a prope

Re: [Live-devel] live-devel Digest, Vol 121, Issue 24

I upgraded the server to the 2013.11.15 version of the code and ran the testRTSPClient built with the same version and the DEBUG_PRINT_NPT flag. This is what I'm seeing: Opening connection to 127.0.0.1, port 8554... ...remote connection opened Sending request: DESCRIBE rtsp://127.0.0.1:8554/stream

[Live-devel] getNormalPlayTime returning negative value

I have a project that implements some FramedSource subclasses to send compressed audio and video. On the receiving side, a subclass of the MediaSink class provides the ability to receive the compressed data and perform playback. The getNormalPlayTime method is called on the receiving side to det

Re: [Live-devel] Presentation time when streaming videorecordingfrom surveillance cameras

Hi Ross Is this more like what you want? Packet no. Time Source Destination Protocol LengthInfo 15312.632819000 192.168.1.43 192.168.1.71 RTSP 224 DESCRIBE rtsp://192.168.1.71/rtsp_tunnel

[Live-devel] Presentation time when streaming video recording from surveillance cameras

ing, it should be possible to get the presentation time from when the video recording was actually made. Any suggestions? Venlig hilsen / Best regards Michael S. Juul B.SC.E.E m...@unitek.dk <mailto:m...@unitek.dk> Unitek A/S Vævervej 5 8800 Viborg

[Live-devel] Frames lost on TS file created with testH264VideoToTranspotStream test

displayed (missing 160 ms What is the reason for the delays in frames ? Thanks, Michael ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] No data from the VND.ONVIF.METADATA subsession

On Sat, May 18, 2013 at 6:09 PM, Ross Finlayson wrote: > > On May 18, 2013, at 2:42 PM, Michael McCandless > wrote: > > I'm using testRTSPClient to pull an RTSP stream, and I noticed the > camera provides two subsessions: > > [URL:"..."]: Initiated th

[Live-devel] No data from the VND.ONVIF.METADATA subsession

I'm using testRTSPClient to pull an RTSP stream, and I noticed the camera provides two subsessions: [URL:"..."]: Initiated the "video/H264" subsession (client ports 56266-56267) [URL:"..."]: Set up the "video/H264" subsession (client ports 56266-56267) [URL:"..."]: Created a data sink for th

[Live-devel] Low Frame Rate Live Streaming

Hello, I am using Live555 to stream raw H.264 live video via RTP/RTSP. In order to provide the video at various frame rates, 1 or more uncompressed video samples are dropped before being passed to the H264 encoder. This works well for the source frame rate of 29.97 fps and also frame rates of

Re: [Live-devel] Range Header

Just wondering if this bug is a priority or if it may be a while. From: Ross Finlayson Sent: Monday, July 23, 2012 6:20 PM To: LIVE555 Streaming Media - development & use Subject: Re: [Live-devel] Range Header Ok. I tried your test program. I tried with scale of 1 and 2. If the client

Re: [Live-devel] Range Header

Ok. I tried your test program. I tried with scale of 1 and 2. If the client sends "Range: npt=30.000-60.000" the server plays starting at 30 seconds into the clip (clip timing information - not wall time) and plays for 30 seconds wall time (which for scale 2 will be 60 seconds of clip time).

Re: [Live-devel] Range Header

Thanks. I don't think I made myself clear though. When I fast forward with a range it doesn't stop at the requested end time (but it does stop a little while after it). When I fast forward with a range it seems to get confused and play to much. From: Ross Finlayson Sent: Monday, July 23,

Re: [Live-devel] Range Header

Thanks for the help. Also, when I play with a range of "now-60" and a scale of greater than 1 (fast forward) it will play way past 60 seconds although it does stop eventually. Any idea why? Thanks. From: Ross Finlayson Sent: Friday, July 13, 2012 6:39 PM To: LIVE555 Streaming Media - develo

[Live-devel] Range Header

Lets say I am 30 seconds into the clip I am watching. If I say play "Range: npt=-60" it will return "Range: npt=0-" and play from the start. When I use "Range: npt=now-60" it doesn't return a Range header but continues from where it left off and stops at 60 seconds into the commercial IF the s

[Live-devel] Looking For Version "2012.06.26" / 1340668800

I made some change to the code but I have no idea what anymore. I need the unmodified version to compare. Does anyone have this version? I realize old code isn't supported or made available but I was hoping someone may have this. It would save me a lot of time. #define LIVEMEDIA_LIBRARY_VE

[Live-devel] Trick Play and the Live 555 Server

I am writing a proxy for VOD servers. I am using the Live 555 server for testing. The version I have is about 1 year old. How can I tell, from the source code, what version I have? If I say play range npt=30-60 at scale 1 it works. If I say play range npt=30-60 at scale 2 it goes way beyond

[Live-devel] Transport Stream

all works flawlessly. Comments? Thanks for your great work on LIVE555 Michael Stamler, CEO Xicore Video Technologies www.xicore.net ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] liveMedia and CMake

having a GUI for configuring my compile environment a big plus, but this might be the case because I am originally a Windows guy ( blame me for that if you like ;-) My 2 cents. Best Regards, Michael Am 17.06.2010 00:11, schrieb Ross Finlayson: Are there any ideological objections against

[Live-devel] UDP framing in RTP

relates to the size of the buffer in OutPacketBuffer and the message size of a UDP message) without causing the "message size too big" errors on the UDP socket. I guess I am missing something obvious , but can't seem to find it. Any help wou

Re: [Live-devel] Source, Frames and Parsers

Thanks for your reply. In fact I have discrete frames arriving one by one in the deliverFrame method, so I will check with the sufficient buffer size of the DeviceSource to avoid truncating of frames. Michael Ross Finlayson schrieb: However I am still unclear on how stream parsers fit into

[Live-devel] Source, Frames and Parsers

. Michael Braitmaier ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] Use RTP Seperately

Hi Ross - Your response to Sean prompts me to ask for clarification for the both of us. You replied to Sean: >If you're streaming MPEG-4 video via RTP, then you must use a RTSP server. It is my understanding that one purpose of RTSP when streaming MPEG4 video is to communicate the all-impo

[Live-devel] MP3FileSource

My application creates a FramedSource from an MPEG-1, Layer 3 (.mp3) audio file and feeds it to an input of MPEG2TransportStreamFromESSource like this: ByteStreamFileSource* audioFileSource = ByteStreamFileSource::createNew(*env, filename); FramedSource* audioES = audioFileSource; MPEG1or2Au

Re: [Live-devel] .m4v / .mp3 Synchronization

I wrote: I have two independent ByteStreamFileSource objects - One feeds MPEG-4 video elementary stream (.m4v) data to an MPEG4VideoStreamFramer. One feeds MPEG-1, Layer 3 (.mp3) audio data to an MPEG1or2AudioStreamFramer. Those framers then each feed a MPEG2TransportStreamFromESSource obje

Re: [Live-devel] .m4v / .mp3 Synchronization

Ross Finlayson wrote: That's correct. The timestamps (specifically, the "fPresentationTime" variable) should be set by each Framer object. These are used to set the SCR timestamps in the resulting Transport Stream. So I'm not sure why this isn't working for you; you're going to have to track

Re: [Live-devel] .m4v / .mp3 Synchronization

Ross Finlayson wrote: Your synchronization problem occurred when you *created* (multiplexed) your Transport Stream from your audio and video inputs. If you used our software to create your Transport Stream (it wasn't really clear from your message whether or not you did), then you should make

[Live-devel] .m4v / .mp3 Synchronization

Hi Ross - I am prototyping a streaming application using an MPEG-1, Layer 3 (.mp3) audio file and an MPEG-4 video elementary stream (.m4v) file as inputs. I am doing this to simulate our actual encoder outputs since they are not yet available. I recorded these files from two different physic

Re: [Live-devel] Modifying testMPEG1or2VideoStreamer to Stream m4e Files

Hi Ross, I just wanted to follow up... I was able to get around the need for an SDP exchange by taking my MPEG-4 elementary stream, packaging it up into an MPEG-2 transport stream, and sending that out over the network instead. It is my understanding that SDPs are specific to MPEG-4. So using

Re: [Live-devel] Modifying testMPEG1or2VideoStreamer to Stream m4e Files

Hi Ross, Thanks for your reply. So If I understand this correctly, the fact that I am using MPEG-4 video means that I have to somehow communicate the SDP description to my player; and for most applications, RTSP is employed to take care of that automatically. I have successfully used testMP

[Live-devel] Modifying testMPEG1or2VideoStreamer to Stream m4e Files

Hello, I am using Live555 under Ubuntu Linux 9.04, streaming over a simple private LAN to a VLC client running under Windows. Just cables and a switch. No router, no firewalls. I modified testMPEG1or2VideoStreamer to stream unicast to a VLC client and it works well. I now want to modify this a

Re: [Live-devel] MPEG-4 Streaming presentation rate

and testOnDemandRTSP streams things at the proper rate. How do I go about checking the file that openRTSP has created? Where do I look to find out why testOnDemandRTSP always calculates 15 fps? Many thanks, -- Michael Barkowski ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

[Live-devel] MPEG-4 Streaming presentation rate

ing the default frame rate in playCommon.cpp which is used by openRTSP. -unsigned movieFPS = 15; // default +unsigned movieFPS = 30; // default This has no effect. Any ideas? What is testOnDemandRTSP Server missing? Just the frame rate? -- Michael Barkowski _

Re: [Live-devel] Compiling on solaris 64 bit -maybe fixed

> know, so I can make the appropriate changes for the next release of > the software. Venlig hilsen Michael Skaastrup Web:www.conecom.dk Systemkonsulent Email: m...@conecom.dk Con E Com A/S Tlf: 36 93 25 57 Hjallesegade 45

Re: [Live-devel] Compiling on solaris 64 bit -maybe fixed

) -64 -r -dn LIB_SUFFIX =a LIBS_FOR_CONSOLE_APPLICATION = -lsocket -lnsl LIBS_FOR_GUI_APPLICATION = $(LIBS_FOR_CONSOLE_APPLICATION) EXE = As i said earlier I am not a developer so this is based on works-for-me info. Kind regards Michael Skaastrup >The usual meth

Re: [Live-devel] Compiling on solaris 64 bit

laris expert (and don't have a Solaris machine available for testing anyway), so someone else is going to have to figure this out for us. -- Hi. I will try to dig deeper. Contacting SUN right now. If anything turns up I will inform you via this list. Thank you for taking time.. If any

Re: [Live-devel] Compiling on solaris 64 bit

libraries and things like that. Thanks for the reply. Hope this gives further info. If I am all wrong please correct me. Kind regards Michael Skaastrup Ross Finlayson Live Networks, Inc. http://www.live555.com/ ___ live-devel mailing list live-deve

[Live-devel] Compiling on solaris 64 bit

pre compiled or instruct me how to compile for 64 bit solaris? TIA Venlig hilsen Michael Skaastrup Web:www.conecom.dk Systemkonsulent Email: m...@conecom.dk Con E Com A/STlf: 36 93 25 57 Hjallesegade 45 Mob: +45 20 31 41 90 5260 Odense S Tlf

[Live-devel] Streaming from network source

hint to the source code section or class would be very nice and helpful. Thanks in advance. Dipl.-Inf. Michael Braitmaier HLRS - Visualization / Video Conferencing University of Stuttgart Germany Website: http://www.hlrs.de/ ___ live-devel mailing

Re: [Live-devel] Stopping RTSP session from server side

Thanks for quick reply. I'm looking for a gracefull shutdown, so that the client will know that the session is closed. From: live-devel-boun...@ns.live555.com [mailto:live-devel-boun...@ns.live555.com] On Behalf Of Ross Finlayson Sent: Tuesday, December 23, 200

[Live-devel] Stopping RTSP session from server side

Hi, Is there any standart way to stop a RTSP session from server side? In RFC 2326 (RTSP) is written that TEARDOWN message can only be sent from client to server. Is there any other way, or closing the TCP socket from server side is enough? Thanks in advance, Michael Mikhlin The information in