LCCS record/playback questions

Hello, everbody!
I have some doubts regarding LCCS session record/playback:
1) LCCS server starts the playback only when the zip file is completely dowloaded from WebDAV? It will takes forever for large sessions, wont it?
2) If yes, what can I do to improve this upload speed? Is there an, or a set of, datacenter storage service with privileged access to LCCS infrastructure that we can hire to decrease this transfer time?
3) I noted that the LCCS does not download the zip file again per each concurrent playback, it appears to have a kind of internal cache. How does it work? Is there any way to improve it's eficiency?
4) Can I choose which shared model to be recorded or not? It's because there are some PODs that does not makes sense for us in ondemand sessions.
Thanks, guys!
Renato A. Ferreira

Thanks, Nigel.
Is there any known way to force a specifc file name to be used in playback process? It's because I saw the room name within the stored file name, and it made me think that I will need to keep or recreate a room with same name for playback. I agree that multiples rooms are the best way to handle concurrent sessions, but, if it's also needed for playbacks, the number of rooms in my account may grow fastly and make some server-to-server operatios gradually slower.
Or maybe I could also automatically rename the file for something like *********_x002F_PLAYBACKROOM_x002F_sessionID.zip and leave only this room created, as the session ID already is a configurable parameter. Do you see any problem in this idea?
Thanks,
Renato A. Ferreira

Similar Messages

Recording beta question - seeking in recorded streams and syncing playback

hi lccs, some initial questions for you as i get started on the beta..
you say audio and video are saved in separate FLVs on the client server. does LCCS ensure that audio/video sync is maintained during playback?
how are the recorded streams delivered during playback? can you seek into these streams? e.g. if i have a recording of a video chat, can i use a timeline to seek forward and back through that video?
and finally - i will want to play back a recorded audio video chat between two users. will LCCS take care of synchronization between these four streams? (two video, two audio) or do i need to build a synchronized buffering strategy to achieve this?

To reiterate Raff's point - when you setup a ConnectSessionContainer with a
PlaybackAuthenticator (essentially, a "playback application"), that
ConnectSession does all the work of synchronizing streams for you. As Raff
says, you can build a timeline and seek within it - ConnectSession's
archiveManager exposes all aspects of the playback to you.
In fact, here's a basic seek bar you could try :
<s:HSlider width="800" maximum="{cSession.archiveManager.totalTime}"
value="{cSession.archiveManager.currentTime}" id="slider"
change="cSession.archiveManager.seek(slider.value)"/>
lastly, as Raff also says, you're not restricted to playing back the exact
application you recorded. If you had, say, audio/video chat plus text chat
recorded, and all you wanted for playback was the A/V, remove the text chat
from your playback application (and delete the chat FLV from the recording
zip, to save on bandwidth), and your playback app should just work. You can
also totally re-layout how your app looks for playback as compared to
"live", or allow your users to manipulate the layout or contents of what's
being played back as they watch the recording. We think this will allow for
some really interesting, innovative applications.
hope that helps,
nigel

LCCS recording (RTMFP vs RTMP/S protocols)

Hey everyone,
So I've read through all of http://learn.adobe.com/wiki/display/lccs/LiveCycle+Collaboration+Service and been reading through the forums (even in the FMS side which maybe relevant http://forums.adobe.com/message/3345179#3345179 and http://forums.adobe.com/thread/736733 ). Since LCCS recording is in beta it lacks the right docs for now.
Could you guys validate/invalidate my assumptions about LCCS recording:
Note: we are requiring Flash 10.0 and link against LCCS swc (flash 10.0) currently
1) Setting <rtc:AdobeHSAuthenticator id="auth" userName="{ourUserName}" password="{ourPassword}" protocol="RTMFP" /> will do the following:
a) cause a P2P multicasting ( which in turn uses UDP and no communication to LCCS service ) between client SWFs, which means no A/V steam recording.
2) Setting <rtc:AdobeHSAuthenticator id="auth" userName="{ourUserName}" password="{ourPassword}" protocol="RTMPS" /> will do the following:
a) cause a TCP through LCCS server which will record the A/V streams just fine.
3) What would I do if we wanted to take advantage of P2P multicasting while recording the LCCS session? Would we have to do something like: http://forums.adobe.com/message/3345179#3345179 ?
4) According to this: http://learn.adobe.com/wiki/display/lccs/09+Peer-to-Peer+Data+Messaging+and+AV+Multicastin g we could do P2P multicasting with Flash 10.1
a) Maybe I'm completely wrong in #1 and #2 and by requiring Flash10.1 for our app, I get recording working while using RTMFP protocol ?
Thanks,
Alex G.

Hi Alex,
Correct - for right now, when you're doing P2P streaming over RTMFP,
recording doesn't work. In order to work around this, we've recommended
either disabling RTMFP or setting StreamManager.maxP2PStreamPublish to 0. In
the latter case, you'd still be using RTMFP and UDP, and so get most of the
advantages.
Answering each of your questions :
1) Partially correct - RTMFP is required for P2P streaming, but not all
RTMFP streaming is P2P. For example, once there are more than 3 recipients,
we automatically switch to hub-and-spoke, since P2P becomes laborious for
the publisher. You can control this threshold using the
StreamManager.maxP2PStreamPublish.
2) Correct. This is one temporary workaround - I'd prefer the 2nd
workaround, using StreamManager.maxP2PStreamPublish = 0, which still allows
the benefits of RTMFP without using P2P. In the next release, even if you're
using RTMFP, once you turn recording on, it will similarly enforce that
everyone stream over hub-and-spoke (without changing from RTMFP).
3) For now, there won't be any way to do P2P streaming with recording
enabled. In the future (likely quite later), we might have time to enable
this, but it's pretty difficult (maybe not even possible), and it's not high
in priority.
4) It's important to separate these issues - RTMFP doesn't require P2P
streaming. In fact, most of the benefits of RTMFP (primarily the fact that
it's UDP-based) work over hub and spoke. What we've seen is that Multicast
P2P (enabled in 10.1, and as opposed to direct P2P) is often not a great
choice for conversations, since multicast introduces more latency than
simply using hub-and-spoke.
hope this helps,
nigel

Adobe Connect 9 Meeting Recording Playback & Audio

Hello All,
I have noticed a strange issue with my meeting recording playbacks.
First Adobe Connect 9.0.2
Windows 7 Platform
I have noticed that after the recording is over, and I run playback to confirm we got what we need...the connected telephone feed (pgi/centurylink) does not come through on first press of play.
The video stream (slides, screen share) will run properly, but there is no sound upon first press of play. If I click the scrubber and drag it back to the open, then the audio will start to play.
I am ok with this work around...however my learners/audience will not be. I am looking for "Press Play and Listen" functionality to keep it simple.
A curious note (these are often good clues, so I figured I would share) The MP3 player pod which is running in the lobby prior to the start of the speaker, is PC sound and it plays back in the recording with no issue. It is only when the telephone feed starts (layout change, and speaker introduction) that the issue begins.
Any thoughts?
Thanks!

As I understand it, it is on the road map for Adobe. Though I've not heard anything other than rumors, so more information would be hard to provide.
I don't know of any third party extensions, other than using something like Camtasia or another screen capture tool.

WVC54GCA Record & Playback problems : Time issues - Playback - AVI conversion

I have gotten everything to work sort of.
Time: The camera time seems to be becoming issue and I am not quite sure why. I have synced it with my pc an it seems still lose an hour. It will sometimes go and try and find a Nttp server even though I havent specified one.
Playbacks: The program for retrieving and playing your recordings. The window never opens to the correct size. I always have to expand it. The slide bar at the bottom (besides not being able to read the font) doesnt display correctly. The speed bar either doesn't speed up the video fast enough for me or doesn't work at all. I am interested in fast forwarding through the recording to see if there is anything worth saving. I am trying to record 12hours per day.
AVI Conversion: The status box of the conversion box doesn't seem to work. When I convert my video I get lots of avi files and have yet to see the full 12 hours of video converted. Why are the data files broken up.
So I think this has the potential for being a great product but I need to know if these minor but essential software issues are going to be resolved. Are there people working on them. Are they of concern to anyone. My use for the camera is to watch my house and record any mischief. But if I am vandalized then I need to produce a tape of the event. I still havent seen that I can do it with this product.
Thanks for listening.

Did the firmware update but still having issues with date and connecting. I am using the software from another camera that works better than the software that came with the camera. It uses a newer version of the recorder/playback which fast forwards correctly and saves and plays back correctly. The timing issue is irritatting since the camera will not do its scheduled recordings if the time is not the same as the pc. Also I had the camera lock up and not allow me access yesterday (before flash). I had to unplug it to get it back. Yes it has a static ip and the router was powered all day.
Thanks

Cisco desktop administrator Services Configuration Multiline, Monitoring & Recording Remove VoIP/Recording & Playback Services

I remove default VOIP Monitor Service in "Cisco desktop administrator>Services Configuration > Multiline, Monitoring & Recording >Remove VoIP/Recording & Playback Services". Now I can't choose in "Services Configuration > Multiline, Monitoring & Recording > VoIP Monitoring Device" Default VOIP Monitor Service. How can I return choice my VOIP Monitor Service(IP UCCX)?

Hi Kalitamih,
This can be regenerated by the Cisco Desktop VOIP Monitor Service.
Please stop the service and start it (do not restart). Let me know how it goes and if it resolves the problem.
Regards,
Arundeep

Intermitent ringing sound on video recording playback

I purchased new 5th gen blue nano from Best Buy on Friday and tried several times but now can report intermittent ringing sound on recording playback. I've posted 3 separate video sections that demonstrate the ringing sound. You can hear here http://www.youtube.com/watch?v=ogAAm64ewj8 this along with seeming poor battery life means I will be returning soon if no response from Apple. Come on guys, this is not up to your usual quality that we have come to expect.

I did call Apple who could not solve either so I did return to Best Buy for a replacement, which thankfully doesn't have the beeping noise.

Local recording playback through desktop app

I have noticed that you lose some functionality in recorded meetings that are played back locally through the desktop application. Most notable is that you can not download a file from the file sharing pod when played back locally. A few customers have asked me about this and I am not sure how to answer them. Is this an intentional limitation for security reasons, a bug, or just a feature that didn't make into the shipping version? Are there any other recording playback features that are disabled in a local playback?
Thanks,
John

I just tested as follows:
1. I enabled on my server the option to allow users to download recorded content
2. I created a meeting, added the file share pod, then turned on recording, turned off recording
3. Went to my Adobe Connect Central, recordings for that meeting, edited the options to allow this to be downloaded locally, then made available for offline and made it public
4. Downloaded the content, ran the recording
5. I was able to select a file in the file pod and download that file to my local drive as normal
It appears to be working fine on my server and client.
Heyward

LCCS Recording - multiple playback of SharedWhiteBoard

Hi,
Is there a proper way to reset the SharedWhiteBoard component between 2 different recordings?
My specs:
Mac 10.6
LCCS 1.5 + FP10.0
Steps to reproduce:
1) Record 2 sessions
2) Playback session1, SharedWhiteBoard plays my scribbles
3) Playback session2, SharedWhiteBoard doesn't repload the same way the audio/video replays
There's a ASDoc on SharedWhiteBoard.connectSession - "The IConnectSession with which this component is associated.     Note that this may only be set once before subscribe() is called; re-sessioning of components is not supported.     Defaults to the first IConnectSession created in the application."
Does this mean playback won't work since I'm resetting the ConnectSession with the 2nd recording ?
Thanks,
Alex G.

Ooops one important note:
I do not reload the SWF, I keep it going by selecting a different recording from webday and hit play... the code ( slightly modified sampleApp that you guys provide ) does the following to try to initialize:
            private function onplaySelectedRecording():void {
                if (cSession) {
                    cSession.logout();
                    cSession.close("closing because docs say I need to do this, yet without closing I can still get this example to work....odd");
                var timer:Timer = new Timer(3000);
                timer.addEventListener(TimerEvent.TIMER, onDone);
                timer.start();
            private function onDone(event:TimerEvent):void {
                Timer(event.target).stop();
                if (cSession) {
                    _archiveID = _archiveIDSelected;
                    //wow what a hack!
                    playbackBar.reset();
                    onConnectSessionContainerPreInit();
                    swb.connectSession = cSession;              // I guess this is what the docs are referring to as "re-sessioning" ?
                    myCamera.connectSession = cSession;
                    simpleNote.connectSession = cSession;
                    cSession.archiveManager.currentTime = 0;
                    cSession.archiveManager.seek(0);
                    //ok lets login and see if we can get both audio and video; normally audio doesn't come back
                    cSession.login();

QT Pro, movie recording, 2 questions

I have QT Pro and a firewire camera.
I can make a movie recording, great !.
Now two questions
1] What is the maximum size of the movie (I want to make one recording of 24 hr)
2] Now the movie is saved to my desktop. Can i change the directory (external harddrive)
Thanks
Jitse

1) The native recording quality of your movie is 320x240 using the H.264 Codec. I have yet to hear of anyway to make it bigger, but you can change the codec to MPEG 4 if you wish in the preferences menu, but the size of the created movie file will still remain 320x240. Once the recording is done, you can always make the playback size bigger by using the view menu or simply dragging from the bottom right corner of the window - but this will not increase the quality at all.
2) Unless you are linking to it in a web page or other application (iMovie or Final Cut Pro), there is no problem moving the file to an external drive. If it is linked, you simply need to update the directory path in your linked applications or web-page.
Hope this helps

ALSA: Recording Playback Audio with HDA-Intel

Hello,
My alsa-info. I only use analog output.
Alsamixer looks like this: http://ompldr.org/vYmE4Mw http://ompldr.org/vYmE4NA
I would like to record audio which is being played back on my computer. I need it for recording of gameplay videos. I want to use only ALSA for this and want to avoid PulseAudio, if possible.
Recording audio playing on the computer with Alsamixer tells me that I should use the "Mix" control of the Capture device. However, HDA-Intel appears not to provide such a control.
Does anyone know how I can create such control? Or perhaps I can somehow fake a capture device, that in fact simply plays back the hw:0,0 playback device?
Perhaps the snd-aloop module is a solution.
I will now start reading about .asoundrc files but wanted to leave this question here, because I feel that I'm likely to fail in my search for knowledge.
Cheers

I can't see a straightforward way to configure icecast to do this on the Arch Wiki.
I tried alsaloop, but it blocks devices when in use. Maybe I'm doing it wrong.
alsaloop -C hw:0,0 -P hw:0,0 -t 10000
I'm not the first to have this problem (thread of 2010-05-22)
I managed to get play-to-file working but ffmpeg doesn't seem to be able to handle the raw output and I'd have to make "default" play to file, as most games don't allow to set what alsa device to use for sound.

Footswitch to control record/playback

It seems to have been a few years since anything similar was posted and since technology moves on.....
I'd like to use a floor switch or button pad to control the playback, rewind and record functions on Logic Pro X. Unless it can also be done by vocal command :-)
Obvious reason is that I've got my hands full of bass, or guitar, or I'm sat at my piano, or standing up ready to sing. What I don't want to be doing is what I'm doing now. Hitting 'R' then waiting for the metronome count to start . I often seem to be unprepared that way.
So what floor switches are currently available that can handle these tasks? The three requirements are obviously Record, rewind, playback.
TIA.
Rob

Gurdas wrote in news:50650000000500000052270100-
[email protected]:
> Eric,
>
> Greetings from India and Thanks! for your help
>
> I have further questions:
>
> 1) The user would press the foot switch for a very small time window
> (say 0.5 sec). What if the program does not read the relevant DAQ
> channel during this time? Thus, by the time the program completes the
> current task and reads the channel, it would find a voltage of 0-1VDC
> inspite of the user having pressed the switch. In the unpressed state,
> the switch will output 0-1VDC.
>
You might be able to put the foot switch read in a high priority thread,
and run it fast.
Another way would be to not use a momentary foot switch, and use one that
toggle
s instead.
A better way would be to use your foot switch to drive a one-shot circuit
with a pulse width that is sufficiently long. This requires a minimum of
hardware design.
> 2) Should I connect it to DI or AI of my PCI-6034 (along with
> CB-68LP)?
>
Either, but I would use a digital line in.
> I have done a lot of AI before but never used an external signal to
> control program flow.
>
> Best,
> Gurdas

Simultaneous record & playback

Been looking all over for something on this but nada, here on Apple.com, Google or in tutorials on line, so:
With track 1 as an audio (real instrument) track playing something previously recorded, I want to record additional tracks with Trk 1 as a backing track, one trk at a time, adding audio BUT, without recording the audio from trk one (the backing trk).
For instance, if I put traffic noises on trk 1, then want to put a galloping horse on trk 2 by itself while cueing by listening to trk 1 (with no audio from track 1 being recorded on trk 2), how can that be done?
Have tried different settings in Track Info with no luck. There's always sound from trk 1.
Want to build some SFX (each FX on one trk) against a backing track, but don't want to record the backing track with each FX track.
Am using GB3 on an MBP (June 07), OS 10.4.11. After the backing track is recorded, I input FX via FW from a Korg Zero8 sounds created on various hardware synths.
I can send from the Zero8 to an outboard amp & monitors and use that for the backer, but isn't there a way to simplify and have the backer in GB, but not audible on subsequent trks?
It sounds like a very basic question, but it seems to have stumped some supposedly adept Logic users, so maybe someone here will have a very appreciated tip.
Thanks!

Hangtime,
Thanks for the rapid reply! have tried two ways: 1) via cans in the MPB headphone port, after setting Audio MIDI/GB + OS prefs set for internals. 2) Reset all prefs to my external Zero8 and from there to studio Mackies. GB set for input on CH 1 &2 Stereo, GB Monitor OFF, no GB EQ or effects.
As noted below to Henri, no matter which routing, the second (or other tracks, pick up audio from the backing track. The backing track gain on the subsequent tracks is about 40% less than on the original backer.
If this is not normal behavior, there is a possible fork in the signal flow that mioght be the culprit which I need to check: I have JackPilot linked to one of the audio deices in AM Setup. Maybe that's causing the leakage. But I when I use that (for other projects, the patch from playback on one track to another armed has no gain loss like the above.
Does the above help?
BTW, I went through all the topics on your (I think it may be yours) tutorial and that's very useful inof and well done...thanks for that effort. However, didn't see anything specifically related to this.

Record playback is very slow between Flash Server and Flash

hi all
I have done server side recording successfully with red5 flash media server and can see
that the flv record is being created properly.
Then if I try to playback the record like
NetStream.play("record.flv"), I see that stream playing is very very
slow, for both of audio and video.
And then if I try to playback the recorded file via using the
filesystem path of it, everthing seems fine.
Problem occurs when I try to sucscribe and play the stream of the record.
PS : There is no problem with playing the published stream
(NetStream.play("streamName"))
Does anyone have an idea about this issue?
If you need any additional information, please let me know.
Thanks in advance
Regards

Hello Fabio,
Just for confirmation.
Is the Integration Services is updated on all VMs on that particular host? IC version.
Could you please compare the vNIC properties/advanced between the functioning VM and the slowest VM network?
All VMs have one vNIC or more than one?
How are you copying out data from within the VM? \\IP Address\ OR \\DNS name\ ?
Is the file transfer issue around the same two VMs?
Could you please double check the Adapters and Bindings priority under
Network Connection\Advanced\Advanced Settings?
Regards,
Charbel Nemnom
MCSA, MCSE, MCS, MCITP
Blog: www.charbelnemnom.com
Please remember to click “Mark as Answer” on the post that helps you, and to click “Unmark as Answer” if
a marked post does not actually answer your question. This can be beneficial to other community members reading the thread.

Recording/Playback on DVD Player

I've already programmed my Verizon remote control to my DVD/VCR player. Which remote control do I use to record and playback. I'm recording from DVD to VCR (VHS Tape). Also, do I have to change TV input for DVD player? TV is currently Video 1. I had DIRECTV and had to set to Video 3 for DVD Player. Any assistance is appreciated. Thanks in advance.
Solved!
Go to Solution.

I called Verizon FiOS Tech Support for help with my dilemma. You have to change the video channel from Video 1 (TV) to Video 3 (Audio/Video) first. First, press the DVD button at the top of the Verizon Remote Control; second, press the TV button also at top; third, press the A/V input button at the bottom of remote control until Video Channel is on Video 3. Use your DVD/VCR Remote Control to control record, play, etc. I hope this helps someone who may have the same problem I had. Good luck!

LCCS record/playback questions

Similar Messages

Maybe you are looking for