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KFI Radio Live Remote: How is it done these days?

If it's not too late to respond to this thread:

* VoLTE is just a proprietary VoIP protocol agreed upon by several cellular carriers, one that has an "HD" (>4 kHz) audio bandwidth. You only hear the added quality when making mobile to mobile calls because the global SS7 PSTN landline telephone network isn't capable of passing anything over 4 kHz with voice calls. So when you call a non-mobile number (like a radio station's call-in lines) from a VoLTE capable cellular device and carrier, the extra fidelity is lowpassed off. I have always wondered why a famous equipment maker like Telos doesn't figure out a way to make a radio call-in system that runs off cellular so the extra fidelity of these calls can actually be heard on-air. Imagine some box that sits in the board op's booth that's connected via coax to a cellular antenna up on the station's roof. Combined with some kind of switching/forwarding service that operated at the provider level, calls from mobiles would come in over that box, while calls from listeners using regular landlines would come in through the station's normal lines.

* Landline isn't exactly analog. Only the "last mile" (the copper between the CO and you) is still analog. Everything else (between all the central offices) has been digital for decades (it began going digital in the 1970s). The digital format used throughout the SS7 PSTN is (I believe) G.711. And I know it's definitely lossless 8 kHz sampling at a 64 kbit/s bitrate, yielding exactly 4 kHz audio bandwidth maximum. Though in practice, most calls only get around 3.4 to 3.6 kHz due to most telephone company backbone trunk circuits (virtual digital circuits anyway) using "robbed-bit signaling." Bit robbing also lowers their maximum bitrate to 56 kbit/s, which is why we topped out at 56k modems back in the day. (Bit-robbing wasn't done on ISDN circuits, which is why ISDN modems were 64 kbit/s -- or multiples of that, when bonded.) Read https://en.wikipedia.org/wiki/Robbed-bit_signaling if you're bored. It also has a nice explanation of where that mysterious, ultra-faint 1333 Hz tone a lot of board ops dealing with callers sometimes heard depending on which circuits their calls traversed, and probably had no clue about the origins of.

* The reason old-style non-VoLTE digital cellular audio sucked (specifically: sounded like crunchy, garbled water being modulated to impersonate human speech) was/is because it used/uses codecs like AMR at astonishingly low bitrates (as low as ~4 kbit/s). The lower the bitrate, the worse it sounded, and from one call/moment to the next, the bitrate would be decided on by each caller's cellular provider based on a combination of factors ranging from network-wide policy to congestion levels at each particular tower. Anyway, you can actually use the well-known open source tool FFmpeg to convert WAV files to AMR and back in order to re-create that exact "classic digital cellphone sound" on a PC. Just google the Windows static binary build of ffmpeg.exe (no installation necessary, you just plop it in a folder and run it at the cmd line), and run it first with this command:

ffmpeg -i input_file_here.wav -ar 8000 -ab 5.9k intermediate_file_here.amr

Followed by this command:

ffmpeg -i intermediate_file_here.amr output_file_here.wav

Replace "5.9k" with your choice of 4.75k, 5.15k, 5.90k, 6.70k, 7.40k, 7.95k, 10.20k, or 12.20k, depending on how bad/good you want the "call" to sound. Great way to torture your friend the radio talk show host, sending him spoken birthday greetings in WAV files or whatever, run through the codec that was every radio personality's mortal enemy. ;-)

Considering the bitrates involved, it is actually amazing AMR manages to produce semi-intelligible speech at all.
WHAT A TOTALLY AWESOME POST! THANK YOU!
 
If it's not too late to respond to this thread:

* VoLTE is just a proprietary VoIP protocol agreed upon by several cellular carriers, one that has an "HD" (>4 kHz) audio bandwidth. You only hear the added quality when making mobile to mobile calls because the global SS7 PSTN landline telephone network isn't capable of passing anything over 4 kHz with voice calls. So when you call a non-mobile number (like a radio station's call-in lines) from a VoLTE capable cellular device and carrier, the extra fidelity is lowpassed off. I have always wondered why a famous equipment maker like Telos doesn't figure out a way to make a radio call-in system that runs off cellular so the extra fidelity of these calls can actually be heard on-air. Imagine some box that sits in the board op's booth that's connected via coax to a cellular antenna up on the station's roof. Combined with some kind of switching/forwarding service that operated at the provider level, calls from mobiles would come in over that box, while calls from listeners using regular landlines would come in through the station's normal lines.

* Landline isn't exactly analog. Only the "last mile" (the copper between the CO and you) is still analog. Everything else (between all the central offices) has been digital for decades (it began going digital in the 1970s). The digital format used throughout the SS7 PSTN is (I believe) G.711. And I know it's definitely lossless 8 kHz sampling at a 64 kbit/s bitrate, yielding exactly 4 kHz audio bandwidth maximum. Though in practice, most calls only get around 3.4 to 3.6 kHz due to most telephone company backbone trunk circuits (virtual digital circuits anyway) using "robbed-bit signaling." Bit robbing also lowers their maximum bitrate to 56 kbit/s, which is why we topped out at 56k modems back in the day. (Bit-robbing wasn't done on ISDN circuits, which is why ISDN modems were 64 kbit/s -- or multiples of that, when bonded.) Read https://en.wikipedia.org/wiki/Robbed-bit_signaling if you're bored. It also has a nice explanation of where that mysterious, ultra-faint 1333 Hz tone a lot of board ops dealing with callers sometimes heard depending on which circuits their calls traversed, and probably had no clue about the origins of.

* The reason old-style non-VoLTE digital cellular audio sucked (specifically: sounded like crunchy, garbled water being modulated to impersonate human speech) was/is because it used/uses codecs like AMR at astonishingly low bitrates (as low as ~4 kbit/s). The lower the bitrate, the worse it sounded, and from one call/moment to the next, the bitrate would be decided on by each caller's cellular provider based on a combination of factors ranging from network-wide policy to congestion levels at each particular tower. Anyway, you can actually use the well-known open source tool FFmpeg to convert WAV files to AMR and back in order to re-create that exact "classic digital cellphone sound" on a PC. Just google the Windows static binary build of ffmpeg.exe (no installation necessary, you just plop it in a folder and run it at the cmd line), and run it first with this command:

ffmpeg -i input_file_here.wav -ar 8000 -ab 5.9k intermediate_file_here.amr

Followed by this command:

ffmpeg -i intermediate_file_here.amr output_file_here.wav

Replace "5.9k" with your choice of 4.75k, 5.15k, 5.90k, 6.70k, 7.40k, 7.95k, 10.20k, or 12.20k, depending on how bad/good you want the "call" to sound. Great way to torture your friend the radio talk show host, sending him spoken birthday greetings in WAV files or whatever, run through the codec that was every radio personality's mortal enemy. ;-)

Considering the bitrates involved, it is actually amazing AMR manages to produce semi-intelligible speech at all.
What a trip down memory lane. I remember when I was working at a local station in the late 1970s, we only bought dedicated lines for the studio-transmitter links/audio and the network feeds. For all remotes, we bought POTS lines. We were in a step-by-step central office. As long as the connection was in the same central office, we had a straight copper connection and get about the equivalent of 5kc line. Our contract engineer was always impressed. Now, if the connection was outside our central office, that was another matter. In those situations our calls went over the old N2 and N3 carriers (the old protocols for a interexchange cables).
 
sdwulfdawg-

Earlier I did not describe these aspects in full: The remote broadcast system is two-way, and the audio returning to the remote site from the station control room is called mix-minus because it contains all the audio that is part of the program EXCEPT the audio coming back from the remote. It is the program audio mix, minus the remote audio.

This way the short time delay of the codec-Internet transmission path does not result in remote talent hearing their voice (returning from the control room) in their headphones with a delay. A really short delay could be OK, but longer delays caused by higher quality systems make it difficult to speak because of the distraction of a delay.

At the remote site, the return mix-minus audio (which contains talk-back intercom from the control room) is mixed with the local microphones. This is what the talent hears, a locally mixed facsimile of the program audio. btw the talk-back intercom is also known as interrupted fold back (IFB), leading to the expression mix-minus IFB.

Because remote talent is not hearing the actual on-air mix, there has to be trust. One mistake can have huge consequences.
A remote is one place where talent should absolutely not say anything they would not want heard on the air.

The talk-back system could be duplicated in reverse, so during spot breaks the remote talent could press their talk-back and be heard on a speaker in the station control room, employing one channel of the stereo feed from the remote site to the station.
Most of the time the control room leaves incoming audio from the remote in cue, and that's how the off-air conversation between remote talent and the control room happens during commercial breaks. Some stations set it up so remote talent can talk to callers during the commercial break, and the calls recorded. This would be done in a disc jockey type remote and duplicate the function of recording calls during songs.

At the remote site the mixing console routing that makes this happen is up to the engineer and the equipment available. I've seen some engineers brilliantly make it work with shockingly simple mixers, and others who were fortunate to have a mixer with all the bells and whistles. When I did remotes, I used a Mackie 1402, which was enough to get the job done but with little margin for anything, like a courtesy feed for someone. At one remote some random media person walked up with his device and started plugging into my Mackie to get a feed. I'm a mellow guy, but I jumped up and slapped his hand, don't you do that!

This touches another aspect of doing remotes- keeping the amount of stuff needed to a minimum. I could riff on about that- no fun making multiple trips into a venue and wondering if your stuff will be gone when you come back with the second load.

The interesting thing is with sufficient mix-minus busses on the control room console (back at the station or network) a number of remote locations can be mixed together on the air seamlessly, without echo or inability for talent to hear each other (at least while on the air). This seems like magic to a radio person from ancient times.

Here's a key point- back then most remote talent could monitor off-air (if the remote was in the listening area). These days more people have delay in the on-air/streaming transmission path (profanity, STL, HD radio, audio processor). They can't monitor off air. Thus, the bi-directional codec goes hand in hand with the new world of radio that is not actually real time but may have as much as 30 seconds of delay from live spoken voice and the audience's ears.

TV had all this figured out decades ago. When bi-directional POTS and Internet codecs arrived, radio came up to speed.
All is built on the concept on mix-minus with talkback.

Except for the most basic units, control room consoles have these features to some extent. Now, in todays' device-oriented world, companies are developing or selling app-based video and audio systems for multi-person shows, such as a morning show with talent in 4 different locations. These systems would have group video, intercom-talkback (off-air), and texting within the show group. And other features to help make great entertainment.

All the above is common knowledge in radio broadcasting now. No secrets were shared.
 
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Curious if anyone here knows what ISDN cost on average in 2020 - 2022? According to one site, I saw an "installation cost of several hundred $$ and ongoing costs of about $60/month". Considering how many AM stations especially are on their knees financially, I'm wondering how many would've just gone satellite/automated 24/7 or even gone dark rather than bear the costs of their hosts and/or some staff getting ISDN lines at home?
I worked with a guy who sent his voice down an ISDN line and device he and the station installed in the mid 90s.

Can you imagine earlier though? Each host takes a Marti home.
 
I worked with a guy who sent his voice down an ISDN line and device he and the station installed in the mid 90s.

Can you imagine earlier though? Each host takes a Marti home.

I clearly remember the first time I made a Zephyr connection to the station. I was doing a daytime test from an empty nightclub in Fort Lauderdale. I was crouching on the floor in the DJ booth. The Zephyr beeped, connected and I heard full fidelity stereo audio coming back from the station on my headphones. Just amazing.
 
I clearly remember the first time I made a Zephyr connection to the station. I was doing a daytime test from an empty nightclub in Fort Lauderdale. I was crouching on the floor in the DJ booth. The Zephyr beeped, connected and I heard full fidelity stereo audio coming back from the station on my headphones. Just amazing.
Ah yes. Set the codec for L3-Stereo, hope it connects, and wait for the band to take the stage... Those things sounded amazingly fine in the day. Yet nothing compared to today's 256k-AAC units. I've also heard that ISDN charges are up to as much as $7 per minute for interstate connections. Doesn't take too long to buy some new IP codecs at that rate.

Dave B.
 
@sdwulfdawg - Why, you're welcome.

@BarryATL - You might find this particularly interesting:

What Did the Old Telephone Network Sound Like? Overview of the Recordings

This Youtuber is an old school phone network phreaker (hacker). As a teenager, he and his friends would explore and map out all the inner workings of the network. They used high quality induction pickups and home-brew telephone hybrids so they could record their adventures directly to cassette or reel-to-reel tape at maximum audio fidelity. In this video, check out 13:48 to 19:40. You will hear examples of how high fidelity pure analog, local phone service could be -- even more than the 5 kHz you mention working with. Imagine having VoLTE on your western electric desk phone in the '70s! Alas, while digitizing the PSTN made long distance calls sound vastly improved(*), we lost the amazing "right next to you in the same room" characteristics local calls to people in your own area could have.

(*) For anyone who wasn't alive then, the video also includes high quality tapes demonstrating how bad long distance could sound in those days -- severe roll-off, phase smear, distortion, static, buzzes, hums, clicks, pops, and layer after layer of faint crosstalk consisting of other people's voices, DTMF sounds, busy signals, and ringing signals. Fun times.

*blows 2600*
 
I've also heard that ISDN charges are up to as much as $7 per minute for interstate connections. Doesn't take too long to buy some new IP codecs at that rate.
We learned that the hard way. Our board op connected the ISDN to IMG to pick up the ball game for 2-3 hours on a Saturday afternoon. But I guess we didn't properly train him to hang it up.

ISDN stayed connected until middle of Monday. Engineering nearly blew a gasket when we got sent the bill.
 
yeoldeschool- Indeed, I think for POTS connections that went through just one central office, you might have been speaking on unloaded pairs. Once I lived near the radio station and the central office was just a few blocks away. I had a phone with an electret microphone, and the station had the Western Electric/Bell speakerphone direct wired into the board. The first time I called in from home, the audio quality was noticed. Actual high end on a phone call, wow. That ended when the phone company went to the digital switches, and everything got what I believe was 8-bit digital at 8 kHz sample rate.

In the beginning digital audio did not play fair. It ransacked analog quality, sometimes in a way that appeared intentional.
Of course, digital audio did bring new creative opportunities and now years later everything is great.

However, we are left with a whole bunch of music from that era containing elements that are low bit-rate and low sample rate.

One music production technique back then was overdubbing full analog fidelity cymbals and high hat on the multitrack to go with low-bit rate and low sample rate snare, kick and tom-tom drum tracks previously recorded. You can hear this on some of the Prince hits. Producers will do anything to make it better.

You can hear the 60's version of this, some otherwise muddy hit records had a tambourine (or other elements) added live during mixing to the master tape. It's weird, the bulk of the song is muddy, but there is a pretty decent sounding tambourine there. That is also how they effectively added more tracks, playing live into the mix during bouncing or during final mix down to master. This is why some re-mixed and re-mastered versions are missing crucial elements.

In a paragraph near the end of this, Rod Argent describes:


In other interviews Argent goes on about this. He is happy and grateful for his career and hearing his songs on the radio, but it still bugs him never hearing the actual version that became a hit. Argent is on the same wavelength as old-school radio people about the "correct" version. :)
 
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DaveBayArea- I remember hearing that some engineers hooked up codec re-boot or disconnect GPIO to an event timer (that probably already existed in the facility) to re-boot or disconnect connections daily in the middle of the night, or another time when remotes were not scheduled. If an engineer did this and later used the codec as an emergency STL, this could bring an unpleasant surprise.
 
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And no non-consensual testing was done on live animals...
HAHAAHAHAHAHAHAHAHAHAHAHAHAHAHA!!!!!!!!!!! I NEEDED THAT SO BAD TODAY, DAVID!

YOU ALWAYS SO ROCK, MY FRIEND!!! It is great to see that someone else lives by the concept of the cup of sarcasm runneth over.....
 
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