Recording Multiple Inputs Over the 'Net? 49
TFGeditor asks: "Thanks to the advice of fellow readers from a previous Ask Slashdot, I now have a PC system optimally configured to produce professional on-air radio programs. Now I have a new problem: my radio co-host and I are in different cities located a few hundred miles apart. In order to give the show a real-time (i.e. 'live') sound, we need to somehow connect us so that we can produce a show complete with co-host banter, real-time interaction, and so on. I want it to sound as if we were both in the same studio. How can we do this? Will Skype or other VOIP applications do this without the result sounding 'tinny' (like a phone connection)? Are there other apps that will do a better job?"
POTS? (Score:3, Insightful)
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You do realize that many, even most, of the group conversations you hear on over-the-air radio are between people who are connected via POTS, right?
Re:POTS? (Score:5, Funny)
Yes, and they generally sound like crap. Yes, you can hear them, but they still sound like crap.
Not if you have a sound-warming tube amp, and gold-plated connectors.
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Only if they're calling in with a normal phone. Telephone isn't as bandwidth constrained as you might think. Most of the quality problem in phone calls stems from the cheap electret condenser element that they use for the mic and the cheap, tiny piezo speaker (or moving coil speaker of grossly inadequate size)....
Back when I was in college radio, we used to do radio remotes with a telephone interface. While it did roll off a bit on the high end, it didn't sound bad by any stretch. That was an interfac
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Perhaps its that T1 that confuses you. To the uninitated, the plug looks an awful lot like a phone jack.
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A POTS line (or VoIP equivalent) doesn't do that. For that requirement, you need a link operating with roughly the same encoding parameters as the resulting combined program, likely something stereo and around 22 khz.
And by the way, that ad hominem was unnecessary you ignorant buffoon.
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What I realize is that when the DJ on the morning show does a contest where he broadcasts all day from in front of a retail store, he's not sending the signal back to broadcast house via a monoral 4khz channel. If he's close enough to base, he's using microwave with a telescoping van-mounted antenna. Otherwise he's often using a fractional T1.
(Nitpick: Monoral? Is that talking with only one mouth? ;-) )
In many cases the DJ *is* using a monaural POTS connection, and in many other case, such as talk shows, hours of conversation are conducted with one or more of the participants calling in from a home or office, on an ordinary POTS line.
It's really, really common that the voice you hear on the radio was routed by Ma Bell before it hit the airwaves.
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But none of that is what the original poster asked for. He asked for a way to connect a co-host in another city such that it sounded
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I was looking for this last aspect, de
ISDN (Score:3, Insightful)
I'm sure there is a better, cheaper digital solution out there. Just make sure you have the bandwidth to handle it.
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The catches are that a. it costs $400 for the basic version (only allowing you a connection to one other user at a tim
Re:ISDN - But what codecs? (Score:2)
VOIP (Score:4, Informative)
Two podcasters that have info about their podcasting technology on their sites are: Leo Laporte (http://www.twit.tv) and Glenn Reynolds (http:/www.instapundit.com).
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VOIP sound quality is very good -- depending on your settings, it's generally far higher quality than POTS (which in turn is perfectly fine for voice). The only problem with VOIP is latency. It's a subtle thing, so whether or not it's a factor will depend on the type of discussion, but it can easily throw off comic timing, and it tends to increase the frequency with which people talk over one another, especially when the conversation has more than two parties.
If the tiny VOIP-induced lag isn't an issue
Re:VOIP (Score:4, Funny)
Cheap solutions (Score:1)
One caution about doing this for a production environment: Make sure your router is stable. I played Feng-Shui(The RPG, not the mystical-furniture-placement-thing) over XFire Monday night, but the damned 2Wire router kept crashing, sometimes after only a couple seconds of operation. Had I been trying to do a radio broadcast, that would have been a ton
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With the right codec, you could use Asterisk, since it's completely designed to do this. Problem becoes finding a soft or hard phone that supports those codecs.
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Personally I'd just grab two copies of Skype, forward the ports to minimize latency, and go at it; the quality and latency is good enough for live broadcasting when it is set up properly (again, with ports forwarded), and the quality between two properly configured Skype clients is significantly better than
Lag and body language (Score:2)
zephyr (Score:2)
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Depends how much you want to spend. (Score:5, Insightful)
I listen to This Week in Tech (twit.tv) every week and they encounter the exact situation you have. The way they deal with it is either with Skype (which sometimes causes breakup of one of the hosts due to lag or traffic), or they use an ISDN connection. The ISDN is the best "pro" solution because it allows good quality audio to be passed across a digital point-to-point connection. No lag, no problems. The only problem is that relatively speaking the ISDN is slow and expensive. However, if you want a reliable, lagless P2P connection there's really no better solution for the cost... your next option is a point-to-point frac T1 which can get really expensive. Of course, it depends on the amount of bandwidth you intend to use.
I do some part-time work in a recording studio where often a member of a band is "remote" (or in one case, none of them live in the same cities). Since we're talking multiple high-bandwidth streams the studio actually has several P2P T1's. The results can be awesome as we get real-time audio down the pipe at very high bit rates and resolutions... and the recording can be mixed in real time just as if the band members were there.
Body language might be a loss though. ISDN is good when you're pushing high-quality audio... but you won't be able to get video down that pipe as well. The best way I can think to deal with it is to use two connections; an ISDN for the audio and use an Internet connection with a webcam so you can each see the body language of the other. It'll isolate the traffic so that they're not tripping over one another, and the video feed seems to be the one you can most afford to lose (due to latency, lag, packet drops and so forth).
I wouldn't recommend trying to do a solution across the Internet unless you can live with an occasional dropout.
Also realize that if you're creating either terrestrial radio or podcasts, you have a certain amount of leniency since the quality is lower by default than HD Radio or Satellite. I'm all for spending what it takes... but there's no need to spend more than you need.
Finally, realize also that no matter what the final bitrate and quality of your finished product, the higher fidelity the original streams you mix together, the better. Higher bitrate and quality will give you "headroom" for compression.
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Not necessarily so. I past decades - It's a looonng time since I worked in broadcasting - it was possible to hire both a 'music' circuit from the remote location to the studio, and a 'voice' circuit, which could be a switched POTS call, to the remote location for the required duration. It's then the responsibility of the telco to give you a decent qua
Some cheap options + warnings (Score:2)
Ekiga [ekiga.org] is what I've been using under Fedora Core 5-6 after experimenting with other options. It's an unencumbered SIP client. Make sure to use an up-to-date version. It interoperates well with MS netmeeting. It's works great for personal use.
Most softphones, including the one above, will allow you to choose the audio codec to use for a point to point call. This is a direct tradeoff of bandwidth to quality. You can get a reasonably high quality signal if you
Hi fidelity voiceIP (Score:2, Interesting)
I developed an application that sends CD quality stereo audio over the internet in real time (one way connection). As input, it takes whatever audio is presented to the input of your sound card (which could be professional microphones, for example) and compresses it to 128 kb mp3 before sending via TCP or UDP packets. TCP requires at least 30% more bandwidth than UDP. For UDP, about 384 Kbits of bandwidth should do, while TCP may need up to 512 Kbits. In UDP mode, some UDP packets are returned to the sender
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I think I found a bug...
compression format (Score:1)
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Think outside the box... (Score:3, Insightful)
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When I edited together a two-camera wedding shoot to DVD for a friend, the cameras didn't have the same timecode, and one of them had to change tapes frequently. I used their on-board audio to sync the images together, then another audio recording from the sound system to replace that (which had to be rate-adjusted due to it being just an audio cassette, so having the camera audio helped to establish sync).
If the cameras
Professional? (Score:3, Informative)
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Q: Hey I want to do something kind of fun with internet radio and maybe a podcast but I need...
A: You suck.
Ventrilo VoIP (Score:4, Informative)
Apparently, Ventrilo also allows different sampling rates, so you might be able to pump through a higher bitrate to make the vocal quality better; however, I've never played with that function, so take that with a grain of salt. The default setting works well enough and doesn't sound like a telephone.
It's also available on several platforms. I run the server on my Sun Blade 100 with Solaris 9, but the three of us use the Windows clients for gaming.
Ventrilo or Teamspeak (Score:1)
Record seperately (Score:3, Insightful)
"The Signal" podcast uses this method (Score:2)
Someone has to say it... (Score:2)
But I guess it's interesting...
Thanks to all (Score:2)
TFGeditor
ISDN (Score:4, Interesting)