[Richard Crowley]

Consider also the Streambox product. Many people are using that for such applications and many others. I work with an organization that uses several of them to feed over-the-air transmitters across the northwest US. https://www.streambox.com/

[Richard Crowley]

I have Vegas Pro 9 and regularly drop XDCAM from EX1 and EX3 cameras, and AVCHD from NX5 into the timeline with flawless results.

However trying to drop files from my Ki Pro into the timeline completely bolluxes the audio. The video is fine, but I get only one channel of audio, and the other is empty. And then when I simply try to play the clip on the timeline, it runs for maybe 30-60 seconds and then that one channel of audio just disappears. You can see the waveform display on the timeline, but the audio disappears, and even the level meter drops to zip.

As far as I can tell from the Sony Vegas website, I have the latest version of Vegas Pro 9, and I have the latest version of Quicktime Pro installed, also. The same files when played on a Mac, or dropped into FCP, have audio playback that is completely normal. I know that Vegas 10 is now available, but spending another $300 feels like throwing good money after bad with all the problems I’ve had with Vegas.

I tried a fix to replace the codec DLL with a version that is supposed to fix that specific symptom, but Vegas crashes with the alternate DLL.

This is another major roadblock to trying to use KiPro in a PC environment. I’m seeing the end of the road where I give up trying to use this combination. Something’s gotta give. Either dump the Ki Pro, or switch to Mac. I’m not keen on either of those options.

[Richard Crowley]

Way too many unknowns. But my first guess would be that your XLR to miniplug cord was not wired properly for the application you were trying to do. It is possible that you ended up recording a DIFFERENTIAL signal on the two tracks (because of the way the cord is wired). And then when playing it back with L+R combined, they cancel each other. Do you still see a signal when you look at the meters (or in an editor?) Try listening to just one or the other channel by itself. Trying the adapter was a good debugging move. But you really need to know more details to arrive at any specific conclusion.

Furthermore, most camcorder miniplug jacks provide “plug-in-power” which is a DC voltage intended to power electret microphones. This DC voltage does not play well with many XLR microphones.

Of course, you are correct that meter motion should never be confused for recording a proper sound track. Recording sound without monitoring on a good set of headphones is exactly like framing and focusing without looking at the vewfinder.

[Richard Crowley]

I don’t know if this might be of any help…
https://www.rcrowley.com/ComClone/SonyCCU.htm

I made my own ClearCom (1st generation) clones and then interfaced the system to my CCU-M3 units.

[Richard Crowley]

AMEN, Brother! I grew up with a soldering iron from grade-school and have pretty much taken it for granted. It is amazing the number of problems that could be easily solved by simple logical debugging and mechanical repair skills. But it does seem to be a lost art.

OTOH, I have made some good “finds” of equipment that people are throwing out for lack of a simple repair. Even whole cameras, etc.

[Richard Crowley]

“But I assume on the dB issue that I’m correct in saying that Sound in real live isn’t measured in the same way as used in a studio (or audio mixer)…”

It is not clear what you mean by that, but if you mean what I think you mean, then it is not a true statement. “Sound” (whatever you mean by that?) is measured the same way everywhere.

“…or as represented on a videocamera (gain, +3dB, 6dB, etc.)”

That is not a true statement, either. When we talk about gain it is exactly the same for any kind of electrical signal, whether it is audio or video or even RF.

[Richard Crowley]

“Remember Blair Witch Project?”

There are stories circulating that the audio track for BWP was either heavily post-processed by Hollywood professionals, or even completely replaced by ADR and Foley. IMHO, additional proof that audio is more important than most producers give credit.

[Richard Crowley]

“You have decibels (dB) used to measure sound.”

That is true for both physical (SPL = Sound Pressure Level) and electrical (voltage, power, etc.) measurements.

“0dB is the value that represents sound where an avarage person with normal hearing, hears nothing. In opposition where 110dB is a lot of sound that could cause hearing damage. So far so good.”

That is a generalization of how dB is used for SPL. Note that it has no direct correlation with how the term “dB” is used for electrical signals or for mixing, etc.
https://en.wikipedia.org/wiki/Decibel#Acoustics_2

“In every respectable editing program you will find an audiomixer with UV meters. These UV meters are standard set to 0dB.
So, as far I’m with the program I assume this setting means that the imported audio is unchanged electronically meaning my sound will be imported with the same strength as it was recorded with my videocamera.”

VU meters in the digital world are labeled “0dB” at the very top. That is known as dBFS (deciBels Full Scale) It means that “0dB” is all the bits on, the very maximum voltage you can possible handle. “When you’re out of bits, you’re out of beer” to paraphrase the old advert.

When we are “tracking” in a music studio or recording original dialog on a video/film set, we use some amount of “headroom” between our average recording peaks and 0dBFS. Because 0dBFS is the absolute maximum, there is no “grace area” (as there was back in the analog age). Some popular headroom offset values are -12dBFS or -20dBFS. The higher value (-12) is more commonly found in more casual, 16-bit production (as with semi-pro equipment, or for “run-n-gun” things like the evening news.) The -20 value is more commonly found in 24-bit sound recording for more formal and controlled productions like feature video/film production, etc.

“Not that my audio has 0dB if it would be measured with a audio meter.”
That is rather an ambiguous statement because several of the words (like “audio” and “measured”) have several different meanings in different contexts.

“So I think to know that there’s dB to measure sound and there’s dB to measure the electrical current to change the strength of the audio wave, as wel..”

Yes, you are on the right track there. And there is also dB used other things, as well like measuring mixing ratios, gain, attenuation, etc.

dB is fundamentally a RATIO. It is not an absolute measurement like voltage or weight or height. It is use most commonly as a ratio when we talk about how much gain you need to boost microphone level up to line level (40-60dB). Or how much attenuation you might get through a long length of cable, or how much high-frequency reduction you would get from hanging thick curtains in your studio, etc.

When people use “db” as a MEASUREMENT, it is meaningless without a REFERENCE. Unfortunately, this reference is frequently omitted. For example, in modern digital use when you see the statement that something has “-10dB” output level, they are probably mean “dBV” where the zero reference point is 1 volt.

https://en.wikipedia.org/wiki/Decibel#Voltage

I didn’t know I was so popular. I am traveling in South Africa (a grueling location scouting trip for a large concert tour. It is definitely NOT a vacation.) Currently in CapeTown and back to Joberg this morning and then 30 hours back to home in Portland. NOT looking forward to that, either.

[Richard Crowley]

It is generally not safe to assume that any particular wireless receiver works with any transmitter on the same frequency or channel. It would be unusual for TX/RX of different brands (or even models) to be fully compatible. It might work in some cases, but I would not consider it to be reliable enough for serious use. There are several additional factors besides simply what frequency/channel is used.

Note further that since a receiver can “hear” only one transmitter at a time, you always need a separate receiver for each transmitter.

To be sure, there are some combinations that work as you are asking. For example there are kits available with a single receiver and two transmitters, one “body pack” for clip-on lav use, and one “butt plug” transmitter for hand-held use. But they are specifically designed so that both transmitters are compatible with the receiver. And you can use only one of the transmitters at a time.

[Richard Crowley]

1) LEVELS – The “headphone level” out of a piece of consumer gear is close enough to “line level” (or at least “consumer line-level”) that it is reasonably compatible. It is the least of your problems. I would start with the output level of the player set for somewhere between 1/3 and 1/2 full scale just to optimize the signal-to-noise ratio (SNR). It is most critical to optimize the player output level (for max signal before distortion) to maximize the SNR. Then adjust the record level to accommodate the output level of the player.

2) QUALITY – The quality of headphone outputs is typically not as good as the proper line-level output. Specifically, they tend to be more noisy as a “line output” because they are optimized for low-impedance loads. It is always preferable to use designated line-level outputs vs. any kind of headphone output. This is pretty much universal for all equipment on the planet.

3) PINOUT – The 3.5mm or 1/4 inch output from a headphone jack has left on tip, right on ring, and ground on sleeve. Whereas a 1/4 inch line-level input on a pro recorder may very well be balanced (differential signal on tip/ring and ground on sleeve. Plugging a stereo headphone output into a balanced line-level input will deliver the difference between the left and right channels. This is almost certainly NOT what you are after. So you must use proper cables/adapters to at minimum break out the left and right signals from the 3.5mm mini-phone plug. Adapters and cables to do this are available at minimal prices, and even at Radio Shack (at inflated Radio Shack prices).

4) STEREO/MONO – You didn’t say whether the cassettes are mono or stereo. If they are mono, then summing the left + right outputs from the cassette playback is probably NOT desirable. Listen separately to the “left” output and the “right” output and select the better one. Note that this preference may change from tape to tape, and even from side A to side B of the same tape cassette.

OTOH, listening to L+R summed (or even better L-R differenced) is a good aid to azimuth adjustment for a mono tape….

5) AZIMUTH – Cassettes are most notable for really really terrible azimuth control, even with premium $10 cassettes and “professional” $2500 equipment. The pinnacle of cassette equipment (Nakamichi Dragon) features a servo motor that CONTINUOUSLY adjusts the azimuth while the tape is playing to maintain optimal azimuth. If you are going for only “transcription grade” transfers from your cassettes, then you can probably get away without bothering with azimuth, using a “walkman” style player. But if you are aiming for any kind of quality, you MUST use something that you can adjust the azimuth for each tape before starting the dubbing. This applies to both mono and stereo tapes, and is even more critical for stereo.

If you MUST use the walkman player, consider the copies to be “working copies” and not archival quality. And by no means discard the original tapes.