Wonderful video. It would be interesting to see how latency stacks up if you’re feeding the Duet via AES and skipping the internal A/D and conversely running a D/A before feeding the input of the PSM- all of which would essentially be happening if you’re using a digital desk.
I am loving these vids. I worked for a regional provider in the Upstate SC in the late 90's early 00's. I am an EE now designing electronics, but I loved working audio. Mixed the TV broadcast for the mega church in Spartanburg. They put in a pretty tremendous 5.1 d&b sound system with PM5D consoles. I really grew to love Yamaha's equipment. We did theatrical also, so mixing in surround was a thing.
Thanks for the great video! 11.4kHz seems really low for an IEM. Is that a big issue in reality? Lots of people say the Lectrosonics sound good, but no one mentions the missing top end.
Yeah, from a hifi aspect 11k is low. But IEM is more about getting a reference and knowing what other musicians and yourself are doing rather than critical listening, so it's typically not an issue Except if the monitor eng is trying to recreate those high frequencies which creates loads of problems with limiting and sound quality of other frequencies.
I posted part 2 at the same time as I posted part 1. Did the link not show up in the video? There should have been a link to part 2 right at the end of part 1 as well as listed in the description for the video
Hi, Dave! Thanks for the great videos you've been churning out... Nice shirt, too! I hail from Goleta, CA and we had a Bob's Big Boy there that I miss now that I'm in NorCal.
Awesome!! Would be interesting to see this type of comparison between wired IEMs like the Fischer Amps In Ear Stick and Behringer P2. As I actually find behringers model to be better built with a smoother knob, easier to open up, more distinct LED indicating the battery life and so on. I would love to see you break down the actual differences between the two in terms of sound and build quality, similar to your great series of the M32 and X32. Thank you for your wonderful videos!!
Each conversion tends to be .4ms to .8ms so converting to digital and back is .8 to 1.5 typically with modern gear. Once in digital, FIR filters and processing can slow things further. I have covered latency in several videos I think on the M32, x32 and QU16
So ive tried PSM 200 and PSM 300. The PSM 200 sounds richer and warmer to me however I'm getting static or RF noises when I play a chord on the digital piano. The PSM 300 doesn't make that noise but has a thinner sound... am I going crazy?? From what I understand the 200 Is analog and the 300 is digital. Is there an Analog warm and rich sounding IEM system that doesn't have the RF noises out there?
They should not sound too different except if you overload them. The digital psm300 will sound much worse when driven into distortion My guess is that if you are hearing drastic differences, the units are most likely being overloaded with too much signal. Pay very close attention to clipping and overload and also make sure the settings are set to avoid any clipping Send less signal level to the transmitters and turn the volume of the belt back up higher often helps
@@DaveRat Hey man, thanks so much for the quick reply. I'm not getting any overload/distortion. I do have the vol knob on the PSM200 down almost all the way..not sure if that's the issue (but my piano has a hot signal so I have to put it down alot. I banged a few cluster chords on the piano just to verify the no distortion.
I would say that your volume on the belt pack should be 50% or higher. Then adjust the input level sent to the transmitter to make anout the volume you want to hear. There may be settings for input level or maybe need a pad. What you want to avoid is sending hot to the transmitter. There also can be overload in the transmitted signal Not familiar with those units but my gut tells me that it should not sound drastically different than the 200
Hey Dave, first of all a big thank you for your excellent instructions. With my band, I now dare to go on stage with my Macbook and audio interface (Tascam 18x08). For me, Reaper is easier to use than any digital mixer I've had before. And now my interesting experience from our last band rehearsal, normally we have a roundtrip of 6.9ms from the interface and 5ms from our in ear solution (Xvive u4) gives 11.9ms, but at our last rehearsal I forgot to reset the audio buffer from my last session from 1024 samples to 32 samples at 96kHz. Now the unbelievable with 1024 samples buffer we could hear ourselves fine, much better than with a 32 samples buffer, how can this be, I suspect it's the phase which is somehow awkward with 32 samples. Dear Dave, could you help me to clarify this. Thank you very much Philipp
@@DaveRat Dear Dave, Thank you very much for your quick reply. I have thought about it again and now put a digital delay with 33ms and 16% dry/wet on all monitor channels. For the first time I can hear my vocals and my guitar clearly in the in ear monitor. My vocal range is bass / baritone and just in this vocal range there seems to be particularly many sound wave overlays in the skull, which are even more striking with in ears.
Interesting it's possible that there's a polarity reverse somewhere in your system and you need to delay it in order to not get the cancellation. Try minimizing the delay and reversing polarity
@@DaveRat Certainly, but unfortunately the wave cancellations seem to be strongly dependent on the frequency, so the delay seems to be the easiest solution for my problem.
If things are in time and in polarity, all will sum well. If it is out of polarity, it will never sum and time shifts will drastically change the the frequencies of cancellations. From what you describe it sounds to be polarity as the main issue and time shifts as the secondary issue. It is not uncommon for sound cards to be polarity reversed. Test polarity once that is correct, then find optimal time shift If increasing time shift makes it better, that is often an indication that polarity is reversed. Have you watched the vids I did on polarity and in ears?
Awesome stuff as always. For the noise off of the front panel jack does your scope have an FFT function to characterize the frequency content of that signal?
G’day Dave, oz rich here, talking to you in good faith (as ever - lols) - Further to your rta mic tests on the headphones, I noticed a difference in results depending on which ear! It occurred to me that that’s because of the slight difference in the formation of my skull, which I believe is common to each and everyone of our species? Bravo to your stratospheric success Sir! Richard Bint, two:3Studios - named in homage to Pythagoras, the famous mathematician and the two of three occurrence (not a number 23 occurrence!)
So why would they filter the wireless pack and roll off that top end? I understand its not really going to be all *that audible, but are we gaining any benefit from rolling that off?
They use a 19k pilot tone that divides the left from the right channel with the left channel below the time and the right channel above the tone at double the frequency for transmitting. Then they freq divide the right channel back down and us the the 19k as the reference middle point. So they need to cut out the 19k from being heard on the output, so they filter it. Of you use the packs mono, they don't need the pilot tone and the freq response can be extended
The Duet system is digital and doesn't use a pilot tone like analog stereo transmissions do. In our case, we needed to fit two decent fidelity signals done a fairly small pipeline, and do so with minimum latency, so the tradeoff was limited audio frequency response in the upper register.
When you plug into the front of the panel does it reduce the high end due to the lack of pilot tone? Because it’s not transmitting an RF pilot tone and it’s capturing the whole noise signature does this account for the difference @5:40?
The front panel will have a vastly different signal path. Whatever the differences are, 9ne train of thinking is that the front panel output should be as identical as possible to the pack output and offer an accurate reference point. So ideally the front panel output would emulate the pack output including any impacts of pilot tone and compander. Clearly both companies attempted to do this as the responses do resemble the pack outputs more so than they resemble the input signal, which does not have the hf rolloff in both, and LF rolloff in the shure. But, neither company was complete in accurately emulating the pack output, though Lectrosonics was close sonically, the noise and polarity appears to be engineering oversights. The shure has the additional challenge of trying to emulate or recreate the impacts of the compander andarticats of analog transmission. Also the missed the mark on the freq response and HF rolloff slope.
@@DaveRat would the appropriate thing to do when using those packs until they are re-engineered or patched with software to flip phase on the output in the console or simply leave it
@@Cletusaz the front panel is out of polarity but the pack is in polarity. There are polarity reversals controls on the pack, so u could rev the polarity on the pack and then send to it rev polarity and all would line up but... Just don't use the front panel outs is the best way. Use a spare pack for monitoring the musicians feed. That way you can get the exact sound
Whats the point of doing all this processing, companding, to the signal? If youre converting to digital at the transmitter, why not just transmit the data, and convert it back to analog exactly as it was sent?
Good question, the dynamic range of wireless transmission. Is very limited. Especially if trying to transmit in marrow bands requires for having numerous wireless channels in the limited rf bands allotted to pro audio wireless. So, manufacturers are faced with a choice, massive compression or loads of noise. To try and deal with that by compressing before transmission and then expanding on the receiver side, to try and maintaing dynamic range and minimize noise. All solutions have issues but a quality compander seems the best fix though it has issues.
As Dave points out, we have a limited pipeline through which to send a stereo signal, so we either have to compand (analog) or data compress (digital) to achieve decent fidelity, low noise, and low latency.
It would be awesome if you could use a lab for these videos. I have to turn my volume all the way up… I love these videos but the irony of Dave Rat having bad sounding UA-cam video is too much.
Interesting, I try and set my voice to be down 4 to 8 db and Max peak at -2. Though some videos I forget, and some vids I need to mess with sound a bit and add a limiter, at least the newer vids should have relatively ok sound.
Thank you Emanuel, I was scratching my head as I know I do screw up the audio in some vids as I get list in the edit world, but was pretty sure I had these newer ones dialed in. I do appreciate guidance on improving.
Wonderful video. It would be interesting to see how latency stacks up if you’re feeding the Duet via AES and skipping the internal A/D and conversely running a D/A before feeding the input of the PSM- all of which would essentially be happening if you’re using a digital desk.
Looking to do a test on that
As a mixer who has a massive investment in Lectrosonics, I absolutely appreciate this level of detailed comparison. Thanks Dave !! --
Awesome and thank you!
I am loving these vids. I worked for a regional provider in the Upstate SC in the late 90's early 00's. I am an EE now designing electronics, but I loved working audio. Mixed the TV broadcast for the mega church in Spartanburg. They put in a pretty tremendous 5.1 d&b sound system with PM5D consoles. I really grew to love Yamaha's equipment. We did theatrical also, so mixing in surround was a thing.
Super cool and thank you!
Thanks for the great video! 11.4kHz seems really low for an IEM. Is that a big issue in reality? Lots of people say the Lectrosonics sound good, but no one mentions the missing top end.
Yeah, from a hifi aspect 11k is low. But IEM is more about getting a reference and knowing what other musicians and yourself are doing rather than critical listening, so it's typically not an issue
Except if the monitor eng is trying to recreate those high frequencies which creates loads of problems with limiting and sound quality of other frequencies.
Great to see this follow up Dave. Looking forward to your findings in part 2!
I posted part 2 at the same time as I posted part 1. Did the link not show up in the video? There should have been a link to part 2 right at the end of part 1 as well as listed in the description for the video
@@DaveRat I was watching off a phone so not sure it comes up like it does on a desktop. Found it OK in the end anyway!
@@RussHollinshead cool cool
Hi, Dave! Thanks for the great videos you've been churning out... Nice shirt, too! I hail from Goleta, CA and we had a Bob's Big Boy there that I miss now that I'm in NorCal.
Thank you Carl! Yes, used to live by Bob's in Burbank. Had the short for a long time!
Awesome!! Would be interesting to see this type of comparison between wired IEMs like the Fischer Amps In Ear Stick and Behringer P2. As I actually find behringers model to be better built with a smoother knob, easier to open up, more distinct LED indicating the battery life and so on. I would love to see you break down the actual differences between the two in terms of sound and build quality, similar to your great series of the M32 and X32.
Thank you for your wonderful videos!!
Great vid! One thing on latency. if you use a digital mixer, what is the latency saving from skipping multiple runds of conversion?
Each conversion tends to be .4ms to .8ms so converting to digital and back is .8 to 1.5 typically with modern gear. Once in digital, FIR filters and processing can slow things further. I have covered latency in several videos I think on the M32, x32 and QU16
So ive tried PSM 200 and PSM 300. The PSM 200 sounds richer and warmer to me however I'm getting static or RF noises when I play a chord on the digital piano. The PSM 300 doesn't make that noise but has a thinner sound... am I going crazy?? From what I understand the 200 Is analog and the 300 is digital. Is there an Analog warm and rich sounding IEM system that doesn't have the RF noises out there?
They should not sound too different except if you overload them. The digital psm300 will sound much worse when driven into distortion
My guess is that if you are hearing drastic differences, the units are most likely being overloaded with too much signal.
Pay very close attention to clipping and overload and also make sure the settings are set to avoid any clipping
Send less signal level to the transmitters and turn the volume of the belt back up higher often helps
@@DaveRat Hey man, thanks so much for the quick reply. I'm not getting any overload/distortion. I do have the vol knob on the PSM200 down almost all the way..not sure if that's the issue (but my piano has a hot signal so I have to put it down alot. I banged a few cluster chords on the piano just to verify the no distortion.
I would say that your volume on the belt pack should be 50% or higher. Then adjust the input level sent to the transmitter to make anout the volume you want to hear.
There may be settings for input level or maybe need a pad.
What you want to avoid is sending hot to the transmitter. There also can be overload in the transmitted signal
Not familiar with those units but my gut tells me that it should not sound drastically different than the 200
Hey Dave,
first of all a big thank you for your excellent instructions. With my band, I now dare to go on stage with my Macbook and audio interface (Tascam 18x08). For me, Reaper is easier to use than any digital mixer I've had before. And now my interesting experience from our last band rehearsal, normally we have a roundtrip of 6.9ms from the interface and 5ms from our in ear solution (Xvive u4) gives 11.9ms, but at our last rehearsal I forgot to reset the audio buffer from my last session from 1024 samples to 32 samples at 96kHz. Now the unbelievable with 1024 samples buffer we could hear ourselves fine, much better than with a 32 samples buffer, how can this be, I suspect it's the phase which is somehow awkward with 32 samples. Dear Dave, could you help me to clarify this.
Thank you very much
Philipp
Just use the setting that works
@@DaveRat Dear Dave,
Thank you very much for your quick reply. I have thought about it again and now put a digital delay with 33ms and 16% dry/wet on all monitor channels. For the first time I can hear my vocals and my guitar clearly in the in ear monitor. My vocal range is bass / baritone and just in this vocal range there seems to be particularly many sound wave overlays in the skull, which are even more striking with in ears.
Interesting it's possible that there's a polarity reverse somewhere in your system and you need to delay it in order to not get the cancellation.
Try minimizing the delay and reversing polarity
@@DaveRat Certainly, but unfortunately the wave cancellations seem to be strongly dependent on the frequency, so the delay seems to be the easiest solution for my problem.
If things are in time and in polarity, all will sum well.
If it is out of polarity, it will never sum and time shifts will drastically change the the frequencies of cancellations.
From what you describe it sounds to be polarity as the main issue and time shifts as the secondary issue.
It is not uncommon for sound cards to be polarity reversed.
Test polarity once that is correct, then find optimal time shift
If increasing time shift makes it better, that is often an indication that polarity is reversed.
Have you watched the vids I did on polarity and in ears?
Awesome stuff as always. For the noise off of the front panel jack does your scope have an FFT function to characterize the frequency content of that signal?
It does though I forgot to look. I am guessing it is the ramp up in HF we see on the smaart transfer function measurement of the front panet
G’day Dave, oz rich here, talking to you in good faith (as ever - lols) - Further to your rta mic tests on the headphones, I noticed a difference in results depending on which ear! It occurred to me that that’s because of the slight difference in the formation of my skull, which I believe is common to each and everyone of our species? Bravo to your stratospheric success Sir! Richard Bint, two:3Studios - named in homage to Pythagoras, the famous mathematician and the two of three occurrence (not a number 23 occurrence!)
Great to meet you Richard and thank you!
So why would they filter the wireless pack and roll off that top end? I understand its not really going to be all *that audible, but are we gaining any benefit from rolling that off?
They use a 19k pilot tone that divides the left from the right channel with the left channel below the time and the right channel above the tone at double the frequency for transmitting. Then they freq divide the right channel back down and us the the 19k as the reference middle point.
So they need to cut out the 19k from being heard on the output, so they filter it.
Of you use the packs mono, they don't need the pilot tone and the freq response can be extended
@@DaveRat Cant say i quite understand the purpose and actual functionality of the pilot tone. Where can i learn more on that?
Searching pilot tone rf transmission should get you I to that rabbit hole.
Here is a start
en.m.wikipedia.org/wiki/Pilot_signal
The Duet system is digital and doesn't use a pilot tone like analog stereo transmissions do. In our case, we needed to fit two decent fidelity signals done a fairly small pipeline, and do so with minimum latency, so the tradeoff was limited audio frequency response in the upper register.
Thank you for jumping in and awesome to get info from Lectrosonics directly.
every time with U i'm learning something good ! bravo encore 🙌🏻
Thank you Laurent!
Is the Duet system on v1 or v2 firmware? (Curious since they changed the audio format in v2)
Hmmm, It was a demo unit from Lectrosonics so I am pretty sure it would be the latest update
Can we see this with a Wisycom setup?
When you plug into the front of the panel does it reduce the high end due to the lack of pilot tone? Because it’s not transmitting an RF pilot tone and it’s capturing the whole noise signature does this account for the difference @5:40?
The front panel will have a vastly different signal path. Whatever the differences are, 9ne train of thinking is that the front panel output should be as identical as possible to the pack output and offer an accurate reference point. So ideally the front panel output would emulate the pack output including any impacts of pilot tone and compander.
Clearly both companies attempted to do this as the responses do resemble the pack outputs more so than they resemble the input signal, which does not have the hf rolloff in both, and LF rolloff in the shure. But, neither company was complete in accurately emulating the pack output, though Lectrosonics was close sonically, the noise and polarity appears to be engineering oversights. The shure has the additional challenge of trying to emulate or recreate the impacts of the compander andarticats of analog transmission. Also the missed the mark on the freq response and HF rolloff slope.
@@DaveRat would the appropriate thing to do when using those packs until they are re-engineered or patched with software to flip phase on the output in the console or simply leave it
@@Cletusaz the front panel is out of polarity but the pack is in polarity. There are polarity reversals controls on the pack, so u could rev the polarity on the pack and then send to it rev polarity and all would line up but...
Just don't use the front panel outs is the best way. Use a spare pack for monitoring the musicians feed. That way you can get the exact sound
@@DaveRat you're a legend Dave! Thank you for the reply and helping me continue to learn more everyday. ✌️🖖
@@Cletusaz 👍
Whats the point of doing all this processing, companding, to the signal? If youre converting to digital at the transmitter, why not just transmit the data, and convert it back to analog exactly as it was sent?
Good question, the dynamic range of wireless transmission. Is very limited. Especially if trying to transmit in marrow bands requires for having numerous wireless channels in the limited rf bands allotted to pro audio wireless.
So, manufacturers are faced with a choice, massive compression or loads of noise.
To try and deal with that by compressing before transmission and then expanding on the receiver side, to try and maintaing dynamic range and minimize noise.
All solutions have issues but a quality compander seems the best fix though it has issues.
@@DaveRat ah... makes sense, I had never even considered that. Thanks for that explanation Dave.
👍
As Dave points out, we have a limited pipeline through which to send a stereo signal, so we either have to compand (analog) or data compress (digital) to achieve decent fidelity, low noise, and low latency.
It would be awesome if you could use a lab for these videos. I have to turn my volume all the way up… I love these videos but the irony of Dave Rat having bad sounding UA-cam video is too much.
Are you sure it's on Dave's end? Because his videos are close to -6dB on my console and sounds great.
Interesting, I try and set my voice to be down 4 to 8 db and Max peak at -2. Though some videos I forget, and some vids I need to mess with sound a bit and add a limiter, at least the newer vids should have relatively ok sound.
Thank you Emanuel, I was scratching my head as I know I do screw up the audio in some vids as I get list in the edit world, but was pretty sure I had these newer ones dialed in. I do appreciate guidance on improving.
doo-ayeee.... DUET! Hahahaha... it gets in your head, don't it.
👍