Recording Multiple Inputs Over the 'Net?

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?"

Comments

POTS?

[oyenstikker]

Get a phone with an audio out, plug it into your soundboard/computer, and call him up.

Re:POTS?

[Spazmania]

Sure, because a 4khz bandpass filter sounds fantastic.

Re:POTS?

[swillden]

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?

Re:POTS?

[Anonymous Coward]

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?

Yes, and they generally sound like crap. Yes, you can hear them, but they still sound like crap.

Re:POTS?

[swillden]

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?

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.

Re:POTS?

[ObsessiveMathsFreak]

Needlessly high tech solution or GTFO.

Re:POTS?

[Spazmania]

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.

Re:POTS?

[dgatwood]

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 interface that I could build with probably $5-10 of parts, not counting the (reasonably good) microphone plugged into it, the metal case, or the power supply.

Re:POTS?

[timster]

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:POTS?

[Spazmania]

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. ;)

Re:POTS?

[swillden]

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.

Re:POTS?

[timster]

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.

Re:POTS?

[Spazmania]

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 as if they were in the same studio. I'm no radio expert, but I know telephony and that ain't happening with a POTS line.

Re:POTS?

[crmudgen23]

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, defeating the 8KHz filter, essentially, to record from phone lines so I could do remote live musical performances. the thing is, these things can be pricey... I found some that were as little as $100-200 and ones as much as $500-1000.... and you generally need at least two.... one on each end. I gave up based on price.

however here are some links I found during my research, including some DIY info.

http://www.sagebrush.com/phontech.htm
http://www.bradleybroadcast.com/2001/telephone.htm
http://www.audiotheater.com/phone/phone.html
http://www.taiaudio.com/right/sales/salescatalog/t elephoneinterface/teleinter.html
http://www.omnicronelectronics.com/PC/Computer_Acc y.htm
http://www.hut.fi/Misc/Electronics/circuits/telein terface.html
http://www.dplay.com/tutorial/Mac2tel.html

hope it helps

ISDN

[Ubertech]

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:ISDN

[denali99755]

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 time), and you'll need an iLok, and b. there can be up to a second of latency. Due to the latter problem, I wouldn't recommend using it in a live broadcast setting, but you should be able to edit out the latency and keep it sounding natural if you're putting a podcast together for later distribution.

Hope this has been at least somewhat helpful--this is a fairly exciting new technology that's just making its way into commercial voiceover production, and once they iron out the latency issue (it is kind of annoying, especially for actors trying to do dialog) I could see it saving everyone a lot of money by doing away with ISDN entirely. And good riddance.

VOIP

[rlp]

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 (http://www.twit.tv) and Glenn Reynolds (http:/www.instapundit.com).

Re:VOIP

[swillden]

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 (and it's smaller than the latency introduced by most cellphones, BTW), then VOIP is probably your best, and cheapest option. POTS doesn't have the lag, and the quality is almost certainly fine for voice, but it might be more expensive if you can't find a way to avoid long-distance tolls.

Re:VOIP

[HTH NE1]

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

Ah, you can fix that in post.

timing

Cheap solutions

[Short Circuit]

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 of dead air.

I understand Google Talk supports a VOIP setting, but I haven't played around with it. (Anyone know if that works under Linux?)

Re:Cheap solutions

[Anonymous Coward]

Google Talk VOIP works under Linux with Jabbin or psi-jingle.

Re:Cheap solutions

[reaper]

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.

Re:Cheap solutions

[Guspaz]

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 POTS. Sure, it won't be as good as Speex through Asterisk, but it is a tad easier and faster to set up, to say the last ;)

Lag and body language

[KDan]

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 issue?

Daniel

zephyr

[gEvil (beta)]

Get a Telos Zephyr. Hey, you never said anything about budget constraints...

Re:zephyr

[DavidKlemke]

Well if we're going to go that far I'm sure a HP HALO system would be far more appropriate.

Depends how much you want to spend.

[Thumper_SVX]

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 (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.

Re:Depends how much you want to spend.

[chris_sawtell]

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 quality circuit, and yours to feed the audio to it. you will then be free of distortion and dropout problems caused by them those Intarwebs. You'll only need a microphone, a microphone to line amplifier, and a fader at the remote end.

That's the professional solution.

Some cheap options + warnings

[Outland Traveller]

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 have the bandwidth for it. I'd advise experimentation to find the codec that works best with the resources you have available.

There are some serious downsides to VOIP in general:

- The general internet is not 100% reliable. You will experience clipping and dropped calls at some point. You can mitigate this somewhat by configuring your routing equipment at each end to protect and prioritize bandwidth for VOIP.

- There is usually a audio delay by design for buffering. This may be noticable to a third party.

A more professional setup would install a dedicated line between the two premises exclusively for VOIP, making sure that all routers/switches from end-to-end up prioritize and protect VOIP traffic.

There's almost certainly some commercial endpoint hardware just for this situation, with a selection of professional audio-in/out interfaces for hooking up to your gear.

Hi fidelity voiceIP

[TropicalCoder]

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 to create a kind of handshaking to inform the sender that his packets are being received.

Audio is send four mp3 frames at a time, resulting in a latency of about one-tenth of a second for both send and receive. In UDP mode, there is the option of selecting some number of buffers so that the audio will be buffered to prevent drop out. Of course then lag will be multiplied by the number of buffers. On top of that you have the latency of your internet connection. Altogether, the lag could be quite acceptable if you have a good connection.

This application worked quite well in all my tests, but you could encounter issues with getting past a firewall or a DSL router/modem. Nothing in the software deals with these issues. I would be willing to "permanently lend" this application to you to experiment with, but you would need a certain level of technical knowledge to get past your router/modem/firewall. To use this application, you would mix the incoming signal from your partner with you own voice and music. Your partner of course will be monitoring the show. This software requires DirectX.

I developed this to teach myself about winsock. I don't know if there is any future in this software since it does not employ the RTP protocol for audio transmission and RTSP for audio signaling like a typical VOIP app, and it depends on mp3. However, it works very solidly and efficiently. I have thoroughly tested it both via the loopback on my computer, and over a computer network, with both TCP and UDP. I never managed to find someone capable of helping me test it over the internet. I would be happy to give a copy of this software to anyone wanting to experiment with it, and especially with anyone that has more than my minimal knowledge about resolving these issues like getting past the router/modem/firewall. If some other programmer thinks this may have a future and would like to colaborate with me on some project, that would be great. You will find my email address on my web site - just click on "Contact me" on the main page.

Re:Hi fidelity voiceIP

[radish]

CD quality stereo audio

compresses it to 128 kb mp3

I think I found a bug...

compression format

[ifakemyadd]

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 problematic. (This rings of anti-net neutrality, apologies) Of course you'll be able to get the job done, many VOIP solutions already exist, but getting it done with the highest quality, this is an issue.

Communication related open source on wikipedia has a few options. freespeak is a long standing open source option, created back in 1991 before any voip applications had much taken off. From its website, it appears that you can select compression quality, so this is at least a good first step to check out. But it does warn that irregular pauses occur, from inadequate bandwidth.

Good Luck!

Re:compression format

[flitty]

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.

Re:compression format

[ifakemyadd]

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.

Think outside the box...

[tchuladdiass]

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:Think outside the box...

[HTH NE1]

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 had their internal mikes disabled and recorded no audio, it would have been hell trying to get lip sync right. For a non-video podcast, you still want to keep the conversation's timing close to correct.

Re:Think outside the box...

[Anonymous Coward]

One of my favorite podcasts switched from having one of the two co-hosts recording their skype session, to having them both record their ends, then editing it together. The improvements in sound have been amazing... it no longer sounds fuzzy or metallic. This definitely appears to be the way to go.

Professional?

[rueger]

"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.

Re:Professional?

[potat0man]

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

[WidescreenFreak]

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.

Ventrilo or Teamspeak

[MISplice]

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.

MOD PARENT...

[Anonymous Coward]

... Hilarious!!

Previous-post URL doesn't work

[Anonymous Coward]

Is this the one?

An Affordable Pro-Quality Sound Card?
Posted by Cliff on Thu Sep 28, '06 11:35 PM
Input Devices Music
TFGeditor asks: "The company I work for is launching a pre-recorded radio program. I will be working with other staff (all in remote locations) to create the sound clips and then cobbling the show together (mixing). I will also interface with the co-host at a remote studio over the net via uber-broadband connection, producing our portion of the show as if we were in the same studio interacting with each other. What is the best sound card for the money (PC/XP) for this type of application?"

Record seperately

[Dmala]

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" podcast uses this method

[silicon not in the v]

"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 record their half of the conversation. Editing in post overlays them, and it sounds completely transparent like they're sitting together. Give a listen to an episode of "The Signal" to find out how good that audio and cohost banter sounds.

(subtle plug) On second thought, I think you should listen to several episodes of that podcast to really get a feel for it. You might just also get hooked on Firefly.

Re:ISDN - But what codecs?

[billstewart]

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 channels available you can get 128kbps which leaves you more room for a fancier codec, which is probably more common for music than for voice applications.

Thanks to all

[TFGeditor]

Some excellent recommendations here, as usual. Thanks, all, for the help.

TFGeditor

ISDN

[CokoBWare]

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.

Someone has to say it...

[SanityInAnarchy]

Les is the guy??

But I guess it's interesting...