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According to sennheiser its best to have them on the same band, since they have designed the preset channels to not have any resinat bleed over into other pre programmed channels. But personally I would say, if you have had trouble finding free channels in your area on the band you have then it would make sense to try the other band. If you have plenty of free channels, then i would stick to the same band.

No, this can not be use as an audio interface. i know... its dumb.. but the usb is just for reading the SD card in this thing. This is a discontinued device and many better options are available now. This is more considered a "Field Recorder," thats probably best for bag work like filmmaking, mobile podcasts, or SFX type projects. A portastudio is more of desk work device that works well for live music, studio podcasts, churches etc. But a recorder is a recorder so its just a form factor question mostly. One is better for portable bag work, and one is more intended for desk work. as for what to buy, if you are looking for something similar to this old R-44, look at the sound devices mix pre series, the zoom f series, tascam DR series, etc. but if you want to do live music you will want something with faders, built in filters, and routing options.

There is no built in timecode on this, But timecode is just an audio signal. So if you have a TC generator (like a tentacle, Nanolockit, Deity etc.) you can send the time code audio signal to one of the 4 channels . You will only have 3 Ch left for sound recording though.

@@faucast4948 i figured this out right before you replied. Such a great set up. I’ve upgraded the preamps on this bad boy a few years ago so I didn’t want to say goodbye just yet. You’re a life saver.

i believe it is about 20mW. enough for headphones, but that's about it without an amp of some kind.

@faucast4948 Ok, thanks! So not that powerful at all then

If you are running the sound out of the h5 and and recording on the a6400, and the final file on the sd card from the camera has drift over time, i can only assume it is an issue with the camera. if there are other devices involved like an hdmi encoder going in to a computer there could be a lot of other issues. one test you could do would be to plug a mic directly in to the camera and test if the sound still drifts over time. But either way the only real fix i can suggest would be to try updating the firmware on all of the devices involved. now if you are recording separately on the h5 and a6400 and downloading both cards to a computer and syncing in post, there may be drift over long takes (around 30 min or more.) And this is just a function of using inexpensive consumer gear. Neither devices have true crystal syncing time code so that is just the downside of not spending tens of thousands of dollars on gear. the only fixes there would be to bump the sound a few frames every once in a while on your editing program, and just line it up by eye, or cut the camera and sound more frequently (maybe every 20 min or so) so they realign with each take.

Thank you for replying. I am plugging the H5 directly into the camera. The camera is recording at 25p 100m and the h5 is on wav48khz 24bit. I will try stopping the recording frequently. I was letting it roll to make editing easier but obviously the drift does not help! I actually found there was no drift when they were not connected.

The Nyquist theorem is a calculation of minimum threshold, not the ideal threshold. The higher the sample rate, the more closely the digital signal approximates an analog signal. With voice recording this is not as much of an issue, but in music recording it can produce a fuller more complete representation of the various sounds at higher sample rates, especially if you are trying to record something like a full symphony in to one source. Your brain will do a lot of the heavy lifting to smooth out the sounds at lower bit rates, but this can lead to the listener actually feeling more tired after listening to long sessions at lower bit rates. This is one of the reasons we avoid using MP3 as well. But also what if i am recording music for dogs...

@@faucast4948 Untrue, a 40,000 khz sampling rate can reproduce a sine wave at 20,000 khz perfectly.

@@lundsweden Well if you know you are only reproducing sine waves, 2 samples per oscillation is enough to make the math work, but we are not capturing sine wave, and thats not how the decoders interpret the info. the image here illustrates the concept of the nuance lost at sticking with the minimum. The more samples you have, the closer to the original analog you will get. media.geeksforgeeks.org/wp-content/uploads/20240223131125/image-248.webp

@@faucast4948 Anything more complex than a sine wave with a fundamental at 20 khz would have harmonics over 20 khz... which we can't hear. That diagram is nonsense, most adults can't hear much over 15 khz, let alone 20 khz. Maybe if you're 15 years old you can hear a 20 khz sine wave, by 40 you'd be lucky to hear over 12 khz.

It sounds like you might have the "auto level" turned on since the volume keeps changing oddly, or possibly the limiter is kicking in too hard. i would turn auto level and comp/limiter off if so and try and set a level based on the sound you want to capture. That is hard to do while driving though so maybe bring a friend. also you will probably want to turn on the 20Db pad, since the source is really loud. A good mic will help too. the built in ones on the zoom are not great. but if you are trying to get "movie" sound, we would usually record each sound separately. The Exhaust sound would be while the car is parked (or on a dyno) and the engine is revving, then record separate road noise, and tire squeals, and breaking with very directional mic's (usually a shotgun) and layer them all together in editing. but if you are trying to get live racing sound, we would use a number of mic's mounted around the car, all specifically capturing different parts. there is not really a magic trick though, it will require a lot of experimentation to get it exactly how you want it.

Basically you can think of the compressor and limiter on this are the same feature or plugin on this, just with different settings depending on which you pick. selecting limiter is one preset, and compressor would be a different preset on the same plugin. for the discord thing, the only thing i can think of is it may be using whatever mic you have plugged in, mixd with the built in top mics gong to the other 2 channels.. I would think that would make it sound worse though, but there are a lot of variables there, so its hard to say.

​@@faucast4948I have only 1 xlr plugged. My though is the you have to turn 2nd xlr Chanel to 0 even if there is no mic plugged in, csuse in stereo mod it picks both channels regardless. And in multitrsck it picks separetly.

Depends on a lot of factors. Usually for something like that i would want multiple mics, and i would set levels with specific mics on each instrument, but also try to physically separate each interment as much as possible. You basically need to look at it as recording a whole band, even though you are the only one playing. But also this could be one instance where i might recommend upgrading to one of the newer 32bit recorders. it takes a bit more work in post to mix the files well, but if set in a decent range it will preserve all of the sound information without clipping. Then it just becomes an editing problem.

I assume you mean the Limiter, it helps to keep the level from getting too loud and clipping.

this is an old discontinued model, only available used now. There are many better choices now.

not usually but there are a lot of different types. you will need to look up your specific mic.

When the gain is staged properly we should never need to request a change to performance. Modern microphones make it so the "talent" no longer need to yell at the audience. But this is a constant debate between film and theater people: Projection Vs. Performance.

sure, i dont see why not,

If you are recording yourself playing and have control over everything else, you can set all your levels to not need a limiter, but if its more like recording a concert from the audience i would definitely use the limiter, as well as the -20 pad.