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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?"
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Recording Multiple Inputs Over the 'Net?

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  • POTS? (Score:3, Insightful)

    by oyenstikker ( 536040 ) < minus caffeine> on Wednesday March 14, 2007 @08:43AM (#18345709) Homepage Journal
    Get a phone with an audio out, plug it into your soundboard/computer, and call him up.
    • Re: (Score:3, Insightful)

      by Spazmania ( 174582 )
      Sure, because a 4khz bandpass filter sounds fantastic.
      • Sure, because a 4khz bandpass filter sounds fantastic.

        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?

        • 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.

          Perhaps its that T1 that confuses you. To the uninitated, the plug looks an awful lot like a phone jack.

          • by timster ( 32400 )
            Don't be stupid on purpose. Talk radio shows (the people who have "group conversations") often interview guests over POTS, as you are well aware, so there's no excuse to play like the GP can't tell the difference between a professional rig and a telephone.
            • Re-read TFGeditor's question. He specifically said, "[M]y radio CO-HOST and I are in different cities [...] I want it to sound as if we were both in the SAME STUDIO." Emphasis mine.

              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. ;)
              • by timster ( 32400 )
                Yeah, and I do hope he finds a better solution than POTS. But at the same time, the poster was right to point out that a lot of professional radio is done that way. I'm not sure that a high percentage of casual listeners even notice.
          • 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.

            • I suppose I can't speak for what's common outside the DC radio market where listen. And I won't argue with your assertion that its common for PARTICIPANTS in a talk show to join in via telephone. I listen to NPR now and then and its quite obvious. Nor is it particularly uncommon for one of the DJ's to hit the field on some humorous assignment, communicating back by cell phone.

              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
    • Needlessly high tech solution or GTFO.
    • Actually, I looked into the issue a while ago of connecting phone to the PC's audio interface and found that there's a class of devices used in broadcasting called telephone couplers (or hybrid couplers) that are intended for addressing some of the problems of attaching telephone lines to other audio circuts (namely the line voltage matching, echo cancellation/duplex issues, and also take the 8KHz upper frequency limit of conventional POTS specs out of the equation)

      I was looking for this last aspect, de
  • ISDN (Score:3, Insightful)

    by Ubertech ( 21428 ) on Wednesday March 14, 2007 @08:51AM (#18345799) Homepage Journal
    This may bust your budget, but there are many radio hosts at commercial radio stations who use ISDN lines back to the studio. The digital voice signal is good enough to make the remote broadcaster sound like they are in studio.

    I'm sure there is a better, cheaper digital solution out there. Just make sure you have the bandwidth to handle it.
    • Re: (Score:2, Informative)

      by denali99755 ( 974676 )
      I don't know what your system looks like, but if you have Pro Tools or any software that will run as a VST host, you can use Source-Connect [] to stream broadcast-quality audio from one of your systems to the other. Source Elements, the company that makes the software, claims that you need at least 300kbps down for it to work, although I would recommend going higher than that, personally.

      The catches are that a. it costs $400 for the basic version (only allowing you a connection to one other user at a tim
    • There are two ways to use ISDN for this. The standard way is to just have it be a very clean telephone connection carrying the vanilla telephone audio stream - G.711 8000 samples/sec 8-bit mu-law companded sampling of a 4kHz filtered audio, i.e. regular low-fi telephone audio, but no extra analog-flavored noise and hopefully a decent microphone. The other way is to use some kind of enhanced audio codec, such as one of the 7kHz 48kbps things, and use the ISDN to carry it as data; if you've got two B channe
  • VOIP (Score:4, Informative)

    by rlp ( 11898 ) on Wednesday March 14, 2007 @08:52AM (#18345815)
    I listen to a lot of podcasts on my daily commute. Most use some form of VOIP. Usually sounds fine (as long as they're not doing CPU or Net intensive tasks in addition to VOIP). Some of the podcasts do interviews with non-techy folks in which case they digitize an analog phone line or use VOIP through a gateway (Skype). For off-site interviews, podcasters use various types of digital voice recorders.

    Two podcasters that have info about their podcasting technology on their sites are: Leo Laporte ( and Glenn Reynolds (http:/
    • 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

  • Teamspeak is available for Windows and Linux, and gives you a decent audio quality if you select the right codec. XFire is Windows-only, but sounds decent.

    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
    • by reaper ( 10065 )
      I'm afraid that TeamSpeak has about a 2-second delay in messages being received. Fine for "Watch for the camper on the hill". Bad for just about everything else.

      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.
      • by Guspaz ( 556486 )
        Why would you need a hardphone to use Asterisk? Grab a soft phone that supports a high quality voice codec such as Speex (Asterisk supports it), connect the two Asterisk servers with IAX, and you're set.

        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
  • Won't lag time be a major issue for a co-hosted radio show? I would imagine much of the dynamics of a co-hosted show, and what makes it so much more interesting, come from the immediate, zero-delay interactions between the two hosts. A large part of their ability to interact so quickly is, I would imagine, driven by the "high bandwidth" of communication between them - ie textual (5%), tonal (45%), AND body-language (50%) content... From the sound of it you've done something similar already - wasn't that an
  • Get a Telos Zephyr []. Hey, you never said anything about budget constraints...
  • by Thumper_SVX ( 239525 ) on Wednesday March 14, 2007 @09:21AM (#18346109) Homepage
    If you are really serious about making it sound "professional", then you'll have to be "professional". This means (ideally) a dedicated link between the hosts.

    I listen to This Week in Tech ( 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.
    • If you are really serious about making it sound "professional", then you'll have to be "professional". This means (ideally) a dedicated link between the hosts.

      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

  • I can only speak for free softphones the Linux side.

    Ekiga [] 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
  • 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

  • If you could, I would try to select an application that compresses your 'studio' communication only as much as your 'broadcast' communication. Otherwise, it seems slightly wasteful, because your essentially increasing broadcast bandwidth for 1/2 of your show (the other person) which doesn't even benefit from the added broadcast bandwidth (the compression quality). But the thing about the internet is it's not really designed for high quality real time application, so in general your predicament is problemati
    • by flitty ( 981864 )
      Wouldn't it be possible to record simultaneously on both computers, recording high quality audio, but listen to each other on the phone or VOIP or teamspeak or one of the other options listed here? Then, you can transfer the high quality audio to a single computer and multi-track the high quality audio.
      • Yes you could, but to make this transparent in the broadcast I think would require a good bit of editing. The pauses manifested in the phone conversation will still be present in the recorded masters on each computer. But at that point, it would be nearly as complicated as editing one sound file that recorded the entire voip conversation.
  • by tchuladdiass ( 174342 ) on Wednesday March 14, 2007 @10:04AM (#18346633) Homepage
    If you are only going for the live "sound", but aren't actually broadcasting it live, then you've got a simpler solution. Use whatever quality link you can put up with when talking to your co-host, but don't use that link's output in the final production. Instead, have your co-host also record his session from his end at a higher quality (with only his audio, not yours), and stitch the results together afterwards.
    • Re: (Score:3, Insightful)

      by HTH NE1 ( 675604 )
      Well, still record the crappy audio. It will help to synchronize the separate tracks.

      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)

    by rueger ( 210566 ) on Wednesday March 14, 2007 @10:19AM (#18346851) Homepage
    "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" Hmmm... I remain skeptical, esp. when you're seeking advice from Slashdot. To your question, no, you're not going to use Skype or VOIP for a "professional" broadcast, for any of a dozen reasons. As noted, you need a Telos Zephyr or similar product. There are broadcast quality units designed to transfer audio back and forth over an IP connection, but Skype isn't it. Don't waste time here, check out a few radio trade magazines. [] And, uh, "professional" is much less about gear than about talent and proven broadcast skills. []
    • Translation:

      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)

    by WidescreenFreak ( 830043 ) on Wednesday March 14, 2007 @10:37AM (#18347107) Homepage Journal
    I use Ventrilo every weekend with my nephew about 20 miles away and friend about 500 miles away during our network gaming nights. The sound is really good, it's completely "in conference" where anyone who knows the IP address could join in, and I've never heard the drop-offs or digital skipping that occurs frequently in Skype or Google Chat.

    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.
  • For most of the podcasts I have been a part of we have used Ventrilo as the way of communicating. I am pretty sure you could do the same thing with Teamspeak as well if you wanted to.
  • Record seperately (Score:3, Insightful)

    by Dmala ( 752610 ) on Wednesday March 14, 2007 @12:19PM (#18348721)
    If you both have decent recording capabilities, the best way to sound like you're in the same studio would be to each record your own track. Talk to each other over the phone or VOIP or whatever using a headset, but also speak into a decent quality mic, recording locally. When you start, send a couple of blips over the phone and make sure it gets recorded on both systems, so you have a reference point to sync the files up later. When you're done, just have him send you his file. Load both files into an audio editor, line your blips up to sync them, and you should be good to go.
    • "The Signal" is a podcast about Firefly-related news. That method you mention is what they use, and it sounds incredible. It's also way easy. I had listened to many episodes of their podcast before I was shocked to hear that they are not in the same studio. They described the method in one of their shows. The two cohosts, Les and Kari, are in different cities. They make a call on a cell phone to have the other person's audio in their ear, but then they are just sitting in front of a microphone to reco
  • Some excellent recommendations here, as usual. Thanks, all, for the help.

  • ISDN (Score:4, Interesting)

    by CokoBWare ( 584686 ) on Wednesday March 14, 2007 @06:00PM (#18354879)
    I know of a radio show in Austin, TX that is connected to the radio network located in MN through an ISDN line. It's clean, clear, and digital. I don't know the kind of equipment they use, but it is a direct digital channel between both points, and I would highly investigate this as an option. It may cost money, but it's likely worth it ($50-75/month my best guess). Check your local telecos.

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