Tech: Letting listeners "call in" their own questi

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Tech: Letting listeners "call in" their own questi

Postby The Layfield on Thu Apr 13, 2006 2:58 am

[ADMIN EDIT: This thread has been split from the Who do YOU want us to interview? thread. Kisai made the following comment:

Kisai wrote:So what I think would improve it would be to maybe try and get some of the fans to call in to ask the questions, or pre-record the questions.


to which Layfield responded thusly:]

Good suggestion. Maybe we could get a skype line for people to call in. Theonly problem is I have to find a good way to let the people over skype hear the calls. Having them come in randomly during the cast would be a bad idea i think... we are random enough as it is...
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Postby KillerFish on Fri Apr 14, 2006 12:42 pm

You could get people to pre-record thier questions, e-mail them to you, and then edit them in later.

But that would mean more edititng.

Bah, I still find it wierd to hear my screen name said in an American accent, so sending in a clip with my normal accent would be worse!
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Postby Kisai on Wed Apr 19, 2006 9:45 pm

The Layfield wrote:Good suggestion. Maybe we could get a skype line for people to call in. Theonly problem is I have to find a good way to let the people over skype hear the calls. Having them come in randomly during the cast would be a bad idea i think... we are random enough as it is...


Well, there are three solutions

1. Pre-record, have people email you an mp3 (made with audacity or something that produces GOOD mp3s), with the only requirement being that they have to be intelligble, otherwise just read their question.
2. Live, in order to do this you need to do some tweakage to your setup, probably won't work unless you have a good DSL connection. Anyways, get a second computer and do the stereo-mini-jack to stereo-mini-jack (or if you can pull it off, S/PDIF to S/PDIF) from the secondary outputs to the inputs on the other machine. Effectively you have an analog copy running. You'd also need to have a second Skype setup on that machine. There would be no latency introduced in this fashion, but any noise or cruddy DAC's in the equipment would quickly ruin it. Ideally you'd need a pair of Soud Blaster Live to do it via S/PDIF digital, no noise and as a bonus you can record the entire show using the second machine. However as stated, this would assume that your connection to the internet could handle all the traffic -AND- that nothing goes wrong. Really it's a two-person setup. This also ignores coordination. Anyone that can't stay up to X AM/PM would be left out. So probably the "geek" approach is a little on the overkill side.
3. Semi-Live, Have the questions pre-recorded so the interviewee's also hear them in advance, then just edit the questions back into the audio stream right before they answer the question. Could give them a cue like "Question 4!" and just replace the cue in the audio file with the actual question.


There is also a possiblity of just ShoutCast Streaming it live, and that would then allow the regular skype configuration to work for 'call-in', given you probably would want to disable any soundeffects skype makes as to not interrupt the flow. I have never used Skype so I don't know quite what goes into it's default setup. Just remember that in real Radio Station environments, there is actually a switchboard operator that pre-screens the calls.


Just ideas.
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Postby The Layfield on Wed Apr 19, 2006 11:21 pm

Thanks Kisai, lots of good info. I've actually looked into the two computer set up but I'm hardly tech savvy. Any links to a walkthrough on that? I dont have soundblaster sound cards.. just the ones that came with the computers... (yes i know i should buy sound cards... maybe when KeenCAST is making me some money. lol)
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Postby Kisai on Tue Apr 25, 2006 10:05 pm

The Layfield wrote:Thanks Kisai, lots of good info. I've actually looked into the two computer set up but I'm hardly tech savvy. Any links to a walkthrough on that? I dont have soundblaster sound cards.. just the ones that came with the computers... (yes i know i should buy sound cards... maybe when KeenCAST is making me some money. lol)


Let's see. If you have TWO computers you need:
2 cables with stereo-minijacks 3.5mm on each end
Bestbuy has them for like 8$
http://www.bestbuy.com/site/olspage.jsp ... 7625917379
Best Buy > TV, Home Theater & Audio Accessories > Audio Cables & Connectors > Analog Audio Cables > Product Info

Connect the Line-in to the line-out on each computer. This will give you an analog loop. You then set the mixer to mute the line-in on the KeenCast. So when you want to "get someone in" to the Keencast you just unmute. I'm assuming that skype doesn't echo your own voice.

You'd still either need a second person to act as call screener or something.

(I don't know if this would require a second internet connection, given that skype operates on a P2P priniciple, so operating the second computer would probably cut the bandwidth in half. Maybe if you are using a router/switch instead of a hub it would still work.)

This is actually one of the reasons I don't run skype. My Roomie does, so therefor I'm not sure if two machines behind the same IP address would play nice.


Here's how this should work in theory:

You, the other host and the guest(s) connect to machine 1's line out to machine 2's line in. Anyone connected to machine 2 will be able to hear the conversation from 1.
Next connect the Line out from Machine 2 to the Line-in on machine 1, this will allow all the audio from Machine 2 to be placed in addition to or in place of the microphone.

The only thing I don't know at this point without trying it is if your microphone plugged into one computer will be sent to the other, I know if you unmute your microphone, (you'd have to put up with hearing your own voice) that solves the problem, but it might cause a feedback tone.

... maybe I'll try this with my laptop, linux box(2) and pda and see what happens. I know I have those cables already
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Postby The Layfield on Wed Apr 26, 2006 6:38 am

sweet. let me know the results of your tests.... of course, this should prolly be in its own thread now. lol
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Postby jtdarlington on Wed Apr 26, 2006 9:29 am

Ask and ye shall receive. ;)
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Postby Propellerhead on Wed Apr 26, 2006 9:38 am

Putting it on Shoutcast Is not a very good idea, if there are too many listeners, other people won't get to it (I hate that, and I hate the guy that sits next to me in school)
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Postby The Layfield on Wed Apr 26, 2006 10:41 am

jtdarlington wrote:Ask and ye shall receive. ;)


Thanks for the split, friend!
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Postby Propellerhead on Wed Apr 26, 2006 10:58 am

Propellerhead wrote:Putting it on Shoutcast Is not a very good idea, if there are too many listeners, other people won't get to it (I hate that, and I hate the guy that sits next to me in school)


whoops, sorry, I was thinking about Shoutcast TV at the moment, that's somethin different
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Postby Kisai on Wed Apr 26, 2006 11:04 am

It just dawned on me that both my laptop and the emachine only have three connectors, headphones, mic and line-in. I can buy stereo-minijack Y adaptors that would solve that problem (splitting the headphones instead of using line-out), ugh, trip out into the sun NOOO...

I need to get a soundcard for the third machine. (The PDA so far isn't cooperating with Skype) maybe I'll goto futureshop/bestbuy later and pick one up with
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Postby wrightc on Wed Apr 26, 2006 11:48 am

You're trying to run Skype on a PDA?

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Postby 10.0.0.1 on Wed Apr 26, 2006 11:57 am

On running 2 copies of Skype behind 1 router:
I know nil about Skype but have had more routing classes than I care to recall like and it would depend on how Skype passes through the router.
If it requires port forwarding from the router you are probably out of luck but if it is just a simple P2P connection then it should be handled like any other traffic.
Not sure how that would behave bandwidth wise or if Skype has a documented issue/limit on doing it, but I don
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Postby UnchainedDarkman on Wed Apr 26, 2006 12:45 pm

If you do the call in, I'd love to call in and ask my questions.
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Postby Kisai on Wed Apr 26, 2006 1:19 pm

Okay soo far...

I haven't gone out to buy any additional parts yet, but...

I got Skype working on the PDA via 802.11b/WPA (PocketPC)

I had to create a second account for the laptop, so I think I'm going to have to make 2 or 3 more accounts just to test all this.

The two desktops currently are Gentoo and Asterix@home, since I only have monitor. But since they are right on top of each other I just need to get a sound card for the asterix box (I'm still waiting on the FXO card, so right now it's just idle.)

I was able to get bi-directional communication between the laptop and the PDA with skype, so that solves the "can you run two Skypes behind a router" question, given I changed the default port on the laptop, so that may be why. I can't change the port on the PocketPC however. The question now is can I run three or 4. The wildcard question is the microphone.

I'm waiting for skype to download/compile on the gentoo box, then I'll have enough to test this with.
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Postby The Layfield on Wed Apr 26, 2006 5:07 pm

theres alot of big words in there. I have a desktop and a laptop. i use a handheld mic which seems to work better on my desktop.
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Postby Kisai on Wed Apr 26, 2006 7:20 pm

I'm going to give up on this for a bit:

Theory:

Connecting the line-in to the line-out on the opposite machine and using one as the conference and one as the call-in bridge should work.

Practice:

Depends on your hardware.
My Laptop has a "What U Hear", which makes every sound heard sent as a recording. This is what the call-in bridge needs, or just plain "line-in recording only", you will not be able to use the microphone if "What-U-Hear" doesn't exist or there isn't an alternative like "Mono-mix"
The two Linux boxes I have, use different chips for their audio. No matter what I did with one of them, no audio was sent over skype. However anything sent from the "caller" was sent through. On my other machine (the one that I had the line-out connected to the Line-in on the bridge) this would be the "conference" machine, successfully sent everything to the call-in bridge. Just Skype didn't send anything back to the "caller" (which was the PDA)

This could simply be that the Linux Skype implementation is horribly broken and might work on windows.

Theory:
Two sound cards in one machine

Practice:
I bought a Audigy 2 NX because it was on sale, hooked it up to the Laptop.
The Audigy doesn't have a "What-U-Hear", however what it did let me do is use two playback channels. So I used the laptops internal "What-U-Hear" to stream music (from winamp) to Skype's recording, but had Skype playback through the Audigy. I heard the "hold music" on both linux boxes, but not the PDA. (which was my clue that the second linux box wasn't sending anything.)

My theory here... is simply use a different program for the call-in, given two sound cards. One sound card is the Skype, and the other is the other program. I'm going to try this now.
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Postby STrRedWolf on Wed Apr 26, 2006 7:39 pm

Hmmm maybe a 3 PC setup would work.
  • The Master Handles the Skype connection and sends audio via a patch cable on Line Out to...
  • The Slave which records from the patch cable on Line In, does appropriate mixing for sound fx, compresses, and shoots it out to a Shoutcast or Icecast server. It doesn't get in the way of...
  • The Client who joins in the Skype connection with everyone else.


Also, I used a headset, a Logitech USB Headset 250. This garunteed me a good, crash-free usage on Linux.
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Postby John T on Wed Apr 26, 2006 8:07 pm

Maybe I'm a moron, but is there a reason the Chrises can't just dial-up the caller in Skype and let them join the session for a brief period of time?
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Postby Kisai on Thu Apr 27, 2006 12:14 am

John T wrote:Maybe I'm a moron, but is there a reason the Chrises can't just dial-up the caller in Skype and let them join the session for a brief period of time?


Apparently Skype only allows for 5 users. So if you have Chris Chris and guest, that's 3. Dialing out (at least with skype) makes a lot of noise, though I suppose the noise can be killed.

It's actually looking like the easiest solution is simply to pre-record the questions. Given there is a far-more-complicated-than-necessary way should the KeenCast ever get popular enough to warrant a call queue.

The two-machine setup would work, and theory would allow for 9 users or 19 users depending if they are Intel Windows computers or something else.

There is another solution, which is to use SIP/H323 and not Skype, this is a virtually unlimited way of conferencing, but has the same tradeoff in using skype in the first place, whoever hosts it is going to burn a lot of bandwidth to have it, though at least with SIP/H323 you can just bridge half the callers to one host, and the other half to the other host and only have the hosts bridged to each other. Skype maxes out at 5(non-intel) or 10 (Intel) as a purely arbritary limit.

I figured out the problem on One of the linux boxes, it wasn't setup to be full duplex, hence why the microphone didn't work. however the other computer (the Asterix one) appears to jack the microphone before skype ever gets loaded, which is why it's not working as expected.


Anyway, it looks like pretty much every method is overkill. Sticking to KISS principle I'd suggest pre-recording (and thus solving the pre-screening issue) the questions, just have people send them as mp3's to the keencast email.

Moving up to Live call-in simply requires a "host server" to relay the conversations between everyone. Just the "easy" way to do this is outrageously expensive (ever see those fancy voice conference calls bigwigs in offices have?) and the cheap way is a lot of work to get to work. It also puts off anyone who can't figure out how to get SIP/H323 stuff to work. Skype comes off being more like AIM which is why people use it. I think the Live-call-in is simply defeated by the fact that the podcast isn't live, so there is no real additional benefit to it being live.

Still having people email questions would lead to a lot of inconsistant quality and volume (as in loudness), post-production nightmare. Right now there is enough inconsistancy between the guests and the hosts not-so-quality microphones.

Also to answer the "can you have 4 skype sessions behind a router"
The answer is "Kinda"
What happens is that status disappears or becomes inaccurate once you get a second skype session going. I had a laptop running windows, two desktops running linux and a PocketPC all connected at the same time, with the audio from the three pc's all coming through to the PDA (one analog link between the linux boxes) The audio distortion got pretty bad, though still usable, sounded kinda like medium cell reception.

Anyway, I have one more trick to try tomorrow. If I can't get it to work, I give up.

Not everyone is a silly computer geek and audiophile.

Also PDA skype doesn't support conferencing.
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Postby The Layfield on Fri Apr 28, 2006 9:20 am

im an audio noob... hence the irony of me being a podcast co host and editor...

i might be able to get one of those skype call in lines. I think they cost a bit o dough though. but you can have a number and people can like call in and leave messages i think. Then, maybe i can play the files into my mic so people can hear or something and then edit the call in post show also. I have heard of other podcasters doing this... I can look into it.
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Postby Kisai on Sun Apr 30, 2006 10:06 pm

I'm not sure how you are doing it right now, but from documents found online, there are virtual "mixing" things you can do, all dependant on your sound card.

My laptop for example, you can set "record from" to "what u hear" which would be sufficient to record everything, but it would result in an echo to the other people involved in the keen cast. However this does make it easier to just play stuff at will.

The geek solution that doesn't involve a heavy investment in equipment is to simply hook your iPOD or other mp3 player to the line-in and switch the "recording input" to line-in, and back to mic when done.

some kind of external mixing board is probably better, but we're all over thinking this and ignoring the easist solution of just using what's available.

My audigy 2 nx has a mute button on it, if I put the "call-in" line on that, and the skype call on the onboard, and then plug them into each other, that would solve the call in problem, given the person calling in would need to use something like SJPhone to do a peer to Peer call, or alternatively make the Conference call on SJPhone (just enter the other callers IP address if they aren't behind a router), and the call-in on skype. This is a much simper mechanism for live call in and works for non-live call-in as well.

Having said that, I know -I- could pull it off, but it's probably not simple enough for everyone. Someone just needs to make a conference bridge that automatically records the call that the hosts and guests connect to, therefor no lag. Adding someone in that way is easier (since that's how conference calls work, enter the conference number and PIN, off you go)
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Re: Tech: Letting listeners "call in" their own questi

Postby Stefen10 on Wed Dec 17, 2014 12:13 am

Great website,thanks.


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