Zoom improvements requested by music schools
music.yale.edu
music.yale.edu
I never tried it because I'm not a musician (just a longtime fan of Justin and Winamp), but I always found the concept very interesting. Apparently it is still alive: https://www.cockos.com/ninjam/
Other ideas (this is not new btw), you don't get to hear the other participants but you're all on the same sync'ed clock with a metronome and trust the general output will be ok. Final mix is synced to viewers (obviously on delay to achieve sync).
However, it doesn't stop there. Latency is also important for voice, not just music; excess latency is also the cause of the dreaded zoom fatigue [0].
We humans seem to have brains designed for particular cadences of conversation, and products like Zoom really work to disrupt & disengage these preferences we have, leading to poor communications outcomes.
[0] https://www.wsj.com/articles/why-does-zoom-exhaust-you-scien...
We'll likely fix the latency issues inherent to the way we do live broadcasting on the web today before the human race has the time needed to adapt to the new cadence.
Our vocal communication was solidified into our genetic code over millions of years of iteration. I imagine that in a decade or less, the latency issue will be fixed for most of the connected world.
What servers are you on? I remember hours, days of playing on it, 10 years ago :) But didnt know it was still active!
So there is some sort of hierarchy of instruments or a dumb sync track pre-recorded.
Lets say the delay is 15 seconds, you hear the composite, including your part delayed, and you "think, cool, I played the right thing at the right time, I'll keep going with my same assumptions" or you hear "my part was way off, but the rest sounded decent enough, better do something different."
The part where everyone is just playing terribly isn't a concern or doesn't manifest because you've got a bunch of intermediate or better musicians playing.
This literally is marketed as a speech jammer.
The amount of times I'm tripping up on my own words because the last thing I said is blaring out a friends/family mobile speakerphone and then back into the mic is disappointing on all levels, to say the least.
Anyway, what I'm building is meant more for the repetitive kinds of electronic music, but I solved the problem by just making it work like a shared loop pedal that records up to 16 measures of audio.
Everyone works asynchronously and can add or remove audio on their own time, but loops get synced to other players as they get recorded.
Best part? You can do cool stuff with browsers nowadays (even record uncompressed audio). So it just needs a web browser.
I agree with others, latency will always be an issue over a distance.
Thought Leadership ... ha ha ... I'm impressed by their leadership, I didn't expect latency and timing matters in Music. Yet, I'm also a novice and ignorant :)
10ms only gets light 2/3rds of the straight line distance between LA and NY. After you start accounting latency in infrastructure components, round trips, and the fact that cables aren't laid in straight lines between two users, the idea of live musical collaboration over even short distances starts to seem implausible.
10ms is what I was always told was the generally accepted threshold after which a musician start to experience increased difficulty as a result of the latency. Which largely aligns with my own experience with audio latency. From there the difficulty just keeps increasing, and the performance quality keeps degrading until at some point the musicians and the audience gives up.
I honestly just don't think you are truly appreciating that light is almost a million times faster than sound.
And that’s just for an idealized communication network. A real world use case would have a lot more latency introduced by routing/switching, and all of the typical quality issues you’d expect from ISPs. I used to live within line of sight of the data centre that hosted an online video game that I played, and the lower bound of my ping was about 15ms, it was usually in the 20s. Those factors are why something like this is unlikely to ever provide a high quality experience over even relatively short distances. I’ve just always found it interesting how quickly you start to run into issues with the speed of light when you’re trying to optimize for network latency.
(And no: you are wrong about round trip. If you care about round trip time then 3ms of round trip by sound is so close that the violin player is going to be elbowing the singer. Sound is fundamentally slow, and you need to just accept that.)
Also, the reason you'd care about round trip time, is if you needed mixing or processing to be done to musician foldback (I never mentioned 3ms being an issue btw). The mixing and processing of sound is prior to being sent to foldback monitors is a completely standard process in just about all live and studio sound engineering. Latency is a very important consideration for sound engineers, and it's why in situations like stadium performances, the only sounds that matters to a musician on stage, is the sound coming out of the foldback monitors directly in front of them. I'm getting a rather strong impression that you don't know much about what actually goes into sound engineering.
FWIW, when I asked my professional musician ex about all of this (which was before I read this comment of yours here) she also mentioned monitors, but it was because she claimed a core consideration for her with respect to distance in a performance was volume of her instrument drowning out her playing companions (and when I pushed into that she said if the acoustics were bad enough they would use monitors). She also told me that people she knew were already using software to play with each other over the Internet. If nothing else, I feel like "proof by counter example" should win here vs. your statement that this is just somehow impossible? As others have pointed out in this thread, the SoundJack people exist and claim to do this (whether or not you believe they have users: my ex apparently knows of users, though I don't know if they are using the same software).
https://www.ianhowellcountertenor.com/soundjack-comprehensiv...
Signals are fast, electrons are slow - on the order of a quarter of a millimeter per second.
> the drift speed through a copper wire of cross-sectional area 3.00 x 10-6 m^2, with a current of 10 A will be approximately 2.5 x 10-4 m/s
As I've said elsewhere in the thread zoom is the wrong tree to bark up since these tools exist.
What applications would you recommend? Can it also do video, if not each link in the chain, at least one camera for the conductor (the node providing the metronome audio)? I realize the conductor would hear nobody at all, until playing back a recording from the end of the chain. Playback of said recording could begin after the song, or one measure into the song if there are no tempo/time changes (NINJAM style).
I will try to search around and get you some more examples. I can't remember off the top of my head what else worked that way.
Another technique is to mix all musicians on a server and send the mix back to all of them. Everyone hears the exact same signal, so in theory people could play perfectly in sync, but in practice hearing yourself delayed is very awkward and needs lots of training. We tried it but gave up quickly.
While this approach certainly works from a technical perspective, but I don't see many musical styles where this would yield a satisfying musical result. However, it might work for pieces which are specifically written for this constellation.
Do all of this, and you can collaborate really well with people in the same geographic area as you. Of course you can't get around the speed of light, but you can do a lot better than the default setup or Zoom will give you.
For instance, 22.5ms latency has been achieved at a distance of 900 miles (between the middle of Kentucky and Boston) with SoundJack and this setup:
Both port-forwarded, both using Fast Music Boxes, both on fiber internet connections. First user had an MXL770 XLR mic and an Audient EVO4 interface. Second user also had an XLR mic and an audio interface.
San Francisco is 7 miles wide.
Performers need to learn to ignore the rest and play their instrument while looking at the conductor.
The real problem here is latency, and I'm good friends with the folks who are working the hardest on that issue for real musicians wanting to collaborate with each other in real time. The most important work is being done with SoundJack.
https://www.ianhowellcountertenor.com/soundjack-comprehensiv...
> In practice, musicians ignore sound and watch a conductor when separated by such distances [of 100 feet].
It had never occurred to me that a conductor was a kind of clock at an approximately uniform distance from the performers.
If you are 100 meters away, you have delay of about a third of a second if you use sound, but for light it will be a fraction of a millisecond.
Doesn't matter for the performers, but it matters for the conductor.
I’ve been a hobbyist musician since leaving school 20 years ago. I play piano, guitar, bass, mandolin, cello, drums, etc. All of the typical video platforms fall short of having a low enough latency to achieve “remote orchestra”.
There was one product/service that I tried that got closer than anyone, JamKazam (https://jamkazam.com/) and there were a few audio only options back mid-2000s that had close to 20-50ms which was better.
Got a Unifi AC LR and it only adds about 4 to t 6ms over Ethernet.
This is my understanding as I’m still learning. Fascinating this stuff works at all.
No, it will not. :) You can get to 5ms above the network delay, even using compressed audio (opus), though some sound devices may be picky.
Network delay can be just a few milliseconds even across a city, assuming that no latency murdering devices (wifi or nics in interrupt mitigation mode) are on the path.
This means that you can have lower audio delay from a compressed audio conference crossing your state than is achieved from speed-of-sound delays from a performer sitting on the other side of a moderate sized room!
The future is already here -- it's just not very evenly distributed. (yet)
If you limit your collaboration radius in that way, then sure (though it still depends on the router technology the packets encounter along the way). But if you want to go beyond that, the numbers get bigger. For transcontinental stuff, much much longer.
But thanks for correcting the basic thrust.
This is the same one way delay as sound in air across 12 meters, which would be on the large size for an orchestra pit.
It does get larger as you go further, but live performance with people on the other side of a large continent is completely realistic and has been done. :)
Sure, expecting to go between NYC and someone in china behind the GFW is probably asking too much for a seamless experience. But many people mostly want to work with people in the same country as them ... and for that, with sufficient technology, latency need not be an issue.
In case you're not aware, I'm the original author of JACK, around which JackTrip is built, which is the most likely and reliable tool for such a collaboration. I certainly know people who've done this, and I regret that I didn't realize back in February how useful it would have been to make something like JackTrip into a much easier-to-use tool for computer-naive folks.
37ms cross-country in the USA is optimistic, but certainly possible. US->Europe is not so great.
When the pandemic hit I thought about putting out some pointers to easier to use low latency streaming resources... but... that would require overcoming pandemic lockdown funk. And honestly it's still kind of a technical rats nest to get everything working. It's not exactly musician friendly to need instructions like "next you need to make sure your nic doesn't enable interrupt mitigation when there are more than 100 packets per second...". :)
[I love Jack BTW.]
One of the 2 netjack implementations uses Opus - great stuff.
Absolutely right that the good stuff (i.e. not zoom) requires way too much setup, and even then (as you note) you're not guaranteed reliable function because you don't control all the hops. Soundjack gets the user-side fairly good, but it's still not quite what I think would have really taken off during the pandemic - probably needs a mobile app for that.
And hey Paul, long time no chat!
It sounds do-able. It sounds like we should already be there. Looking around though you’ll see we aren’t because of technical issues described that are systemic of the infrastructure used.
Yes, it will as soon as you are 1000km away. Light speed creates a lot of unresolvable latency. How can you play live between London and Sydney? You need a different musical understanding.
The real problem is computer audio isn’t designed for low latency. Generally, as long as video and audio are reasonably synchronized nobody notices, but latency as rarely a consideration on it’s own.
Even with the state of the art equipment, transcontinental music making will always exclude the kinds of music that most of us play while in the same room together.
AWS latency isn’t great between regions. Looking on some other sites you can see London to NYC is 70ms, but that’s also 5,500km which is far behind the 1,000km mentioned. Ex: https://wondernetwork.com/pings/London
70ms /5.5 = 12.7 ms seconds round trip. The speed of sound is only 1,125 ft/s, and half that for round trips so 1,000km over fiber is the equivalent of ~7.14 feet apart in the same room plus whatever latency your personal computer and software adds.
Well, I could probably get some form of POTS either analog or ISDN/T1
If you use sonic, at least in some areas they have a setting where you can turn down the level of forward error correction and get lower latency in exchange for somewhat higher packet loss.
If your conferencing is 'slowing down' and 'speeding up' then it's simply not a realtime communications medium.
Sure. If you constraint it a little more, you can also just play to the persons in the next room directly (no streaming required).
For country level or worse global collaboration latency will always be a problem however...
He's been teaching music remotely since August, and one of the first days he demonstrated why they have to be on mute on Zoom by having them all sing happy birthday to one of the students. It was utter chaos. I worked with him to try to find a solution, we tried various tools, but found nothing that worked across cable, DSL, WISP, and cellular connections.
Some tools could sync after the fact so the recording sounded good (if each musician keeps the same tempo), but the whole point is that the kids needed to be able to hear and follow each other, and just one student at 500ms latency totally ruins the tech.
So he teaches music to second graders by zoom, and they're all on mute unless he unmutes them one by one.
Even with fiber all ISPs I have ever used at home have latency murdering devices between my modem and their connection to the Internet. And I have to go through these hops even if I connect to an IP on the same IP.
Best I ever got was 12ms. Once you connect your laptop to the Internet in a data center you will instantly know what I mean.
https://news.stanford.edu/2020/09/18/jacktrip-software-allow...
Its also a peer2peer system, and not easy to set up correctly.
Some tests that focus on interconnection issues (ISP to another ISP):
Continuous
To Azure: https://azurespeedtest.azurewebsites.net/
To AWS: https://awsspeedtest.xvf.dk/
One-time
To Cloudflare: https://speed.cloudflare.com/
To Netflix (maybe misleading due to direct interconnectivity in some ISPs): https://fast.com/
There are some good instructions here for different systems.
Even just the difference between local loopback (instrument -> headphones) vs instrument -> USB -> PC -> USB -> headphones, feels like playing with someone continually dragging the tempo down.
I'm sure I could relearn and adapt to >10ms, but this is a far cry from a live experience of playing with others, and I am definitely not alone in my cohort (after trying several low latency remote meetup configurations).
Many of us in classical music are doing this all the time. So it's definitely doable. Especially if you have a conductor! Indeed, that's one of the reasons conductors are useful.
Unless everyone just jam to a click track instead of each other, it's very hard to play together without seeing the body language.
I'm a musician and I'm incredibly confused by this point. Music instruction cannot possibly be effective without a visual component, regardless of whether it's a solo instruction or a group. An A/V method with a bit of latency is almost always going to be superior to audio-only.
I suppose blind people will never be successful with music.
Blind people are perfectly capable of learning to play music. Blind people and non-blind people alike can learn on their own, or seek out instruction. It is the latter that we are discussing here.
Even just touching strings in different ways can produce wildly different results; think playing a harmonic note on a bass guitar, which is how you get such a low-sounding instrument to play a high-pitched note. How you blow air through your horn can greatly change the timbre and mood of your music.
There are a lot more examples, and they are all difficult to correct if you aren't able to see what the musican is doing.
The old analog POTS phone system wasn't like this. It had that feeling of almost being in the same room, no latency, true birectional conversation.
I was wondering if there's anything that tries to get close to this old POTS level of quality, at the cost of higher bandwidth and processing power.
Can some of the software mentioned in this thread achieve this?
To make it work well you Amy have to turn off lots of helpful processing that modern systems do.
I mostly agree with this, but POTS starting going digital surprisingly early, before VoIP ever became a thing. And of course digitization back then was typically chopping a huge amount of the high frequencies, making things sound...well, like the phone.
Most people making calls today have no conception of how good telephones used to be.
In other words, they don't want a real time music streaming app, they want a music teaching app. Duh!
This makes sense if you are trying to teach, you need to see the techniques not just hear them.
Which may be tied to its low latency... I assume there are algorithmic reasons why cancellation needs latency.
That and between the audio setup wizard and its unorthodox auth system, the onboarding process is brutal for non-technical users.
I have defined four scenes in OBS which she can switch with A, S, Z and X, followed by a space bar press.
A is the webcam of the notebook, which sits right above the keyboard of our digital piano. This shows just the semi-portrait of the person playing.
Z shows the view from a second webcam mounted above the keyboard to display the keys and the fingers.
X is the view of just the virtual keyboard from a Synthesia window, where the pressed keys are highlighted.
S is all three previously mentioned scenes combined in one nicely arranged view.
Audio from the digital piano is also a source for scenes in OBS.
OBS provides a virtual camera that can be used like a normal camera in most, but not all conferencing software. For example, the preinstalled Skype in Windows 10 doesn't recognize the virtual cam, but Skype Desktop from skype.com does.
It's not perfect, but - I think - a decent setup, quickly built with only a laptop, one extra webcam and free software.
My better half teaches online as well. Your setup goes the extra mile and leaves all other setups in the dust. Most parents cannot be bothered to make sure a basic setup is working.
For music education even a $5 lav mic would be a huge help.
Now I wonder about performing together? Everything I've seen that looks like a live Zoom performance of 10-20 musicians is actually recorded separately (with a click track) and then edited together in a video to make it look like it's Zoom.
But now that the music quality is there... Zoom could conceivably broadcast a click track (e.g. the host's audio that would go to performers' speakers but not the audience's, or even the host being a conductor or lead performer), performers would play "solo" (without hearing each other), but then the central Zoom server would wait for and cache every performer's audio until they were all received, with a suitable buffer, and then output it mixed, to the audience, synchronized to the host's (click-track) audio (timecodes).
And unlike normal Zoom calls where it only ever transmits 1 or 2 audio streams (the loudest ones at the moment), it would always mix all of them.
It's probably complicated enough that it wouldn't be worth Zoom's effort, and also niche enough not to be worth a startup developing a separate product for it... but it certainly seems doable, no?
It's open source: https://github.com/gwillen/solstice-audio-test
Here's an example with some informal singing my friends and I were doing this evening: https://echo.jefftk.com/recordings/2021-01-15-023012.wav (it makes ephemeral recordings)
This was pre-mobile, so it was expected that you could play multiple videos at once. I was amazed, and it was the intent, that independent of how few or many videos you played and no matter when you started each of them, all sounds blended together nicely.
I don’t know what to call it when there’s no inter-sound coordination - a timing independent harmony? It was so cool.
Anyways, I would be super interested in attending a zoom concept of that sort that might side step some technical difficulties.
We also played a specially composed piece for playing over zoom. We each recorded separately then sent our files in later. But it was live. And the music was super-simple.
Here are a couple of examples I know of on YouTube where a single musician plays multiple parts, which are then edited together:
Now professional cellist, graduating senior at time of recording, Sarah Chaffee playing an original transcription of Alice Cooper's Poison with 5 cello parts (with a short guest appearance by Shostakovich at about 2:42, then showing up again in the outro to play us out at 3:33): https://www.youtube.com/watch?v=PjYwQHFsld0
Professional cellist Wells Cunningham playing the Handel- Halvorsen Passacaglia: https://www.youtube.com/watch?v=NseBdxfHk5k
This guy has a really, uhm, interesting violin technique. ;) Just for the fact that he chose to play the instrument the way he did tells me he does not actually know how to play violin. What he's doing is playing it like a teeny tiny cello. Since the notes are so much closer together on the respective fingerboards on a violin than on the cello, I'm betting he needed an extra take or 2 on that. His performance is passable, but nothing when put up against the performance of an actual, professional violinist.
This can be repeated as many times as wanted. On 3rd time you will hear what others played during the 2nd round and that is presumably better played than what they played the first time when they only heard the click-track.
Thus you can "jam" with the other musicians the way they were last time around.
During the first lockdown I watched a demo video of a remote VST plugin combo that allowed the recording artist to record his vocals, and the recording engineer to manipulate the audio each from their own homes.
There is an iOS version available now too, which has inbuilt synths and effects but also allows external sources. That in essence means people are doing exactly that!
The app is alot of fun even with little musical ability. With studio it seems they're trying to achieve your vision but in the DAW.
There’s a bunch of similar bits of software but none of them are super welcoming for the less technically inclined. I guess that the experience is often going to be limited by network performance (specifically jitter and loss on UDP) so it’s a challenging thing to make super slick/consistent. I think that low latency 5G with latency guarantees might help solve it when its widely available though.
If you have special software like Jamulus, non-wireless internet, players geographically close to each other, and an external audio interface, the latency is surprisingly good. If you have most but not all of that, it can still be usable. Our end-to-end latency was around 30 ms, so not as low as parent's, but we were still able to play even "jam songs" where just followed each other for chord changes instead of having agreed them beforehand.
- ethernet instead of wlan
- dedicated audio interface, takes calculation overhead off your computer
- a proper microphone, improves acoustics greatly
All of these steps will increase Zoom/Skype experience, too. However, Zoom/Skype will hardly go below 100s of ms of latency, just by themselves.
Yeah, I'm skeptical. If you had a child, you'd be so spread thin with keeping your kid healthy and safe, while paying for it all, that the licensing of some software they use would be the least of your concerns. You'd be worried that his girlfriend might be manipulative, if he's smoking too much weed, if he's partying too much, if he is being bullied, or if saving up for the PS5 is a wise use of his summer job funds, most likely.
My worries is while I'm working with one kid the other will start watching Thomas the train or the like on youtube.
For the last few years, there have been complaints that Youtube Kids has issues where the algorithm stumbles onto content using cartoon characters in ways that are disturbing for young kids.
Plus they run around acting like the naughty diesel 10. Instead of imanginative play.
I regularly stumbled upon open jamulus rooms which some music teachers used for their private lessons. They usually spend a lot of time (painfully so) explaining the students how to connect everything and use Zoom simultaneously for video.
I do wish there are more examples for other types of instruments or even voices. Guess i'll have to try it out myself.
Er, sort of. I think there’s some incorrect information in this comment.
Piano is an acoustic instrument, and to record an acoustic instrument well, you generally need to have a decent instrument in a good acoustic space with good microphone placement. For most people, getting all three is quite hard. Microphone placement for piano ain’t exactly easy but it’s not really harder than, say, guitar or violin. The hard part is that you inevitably capture a lot of ambiance from the room. With a room that sounds bad or with poor mic placement you’re going to get an inferior piano sound.
What makes, for example, guitar easier is really just the fact that if you have any decent room somewhere, you can fit somebody with a guitar into it. It’s small & portable. You still need a good instrument, a good room, and good mic placement but you don’t need to have the guitar permanently set up.
The frequency range of a piano & its harmonics are not especially difficult to tame in music production.
There are some instruments that are kind of “easier” in a sense, like vocals, electric guitar, or electric bass—but I don’t think the differences are all that stark. The main difference is that you are much more likely to use only a single microphone for these instruments rather than multiple mics, and you’re more likely use multiple mics for piano / acoustic guitar / violin / etc.
If there’s one common instrument that requires the most effort in music production I’ll say that it’s the humble drum kit, hands down. If you’re in a home studio, the drum kit might spur you to add more channels, make you work harder setting up the instrument and mics, and make you do more work with acoustic treatment.
Now _playing_ together is much more challenging, and not something especially viable for students (which is why it's a bit weird targeting this at Zoom): you definitely need a proper audio interface, most likely with ~5ms of latency. You then also absolutely have to be on a wired network connection, which on most modern laptops probably requires another adapter, as wireless introduces too much latency. Overall you need < 25ms of latency in order to be able to play "live" together. This is doable but you need the right tools.
I'm a folk musician who plays in pubs and you can even see the effects of latency in a real-life setting like that - if the room is too large then people on the other side start to drop out of time. What I do in this situation is follow their hands rather than what I can hear. Unfortunately this too doesn't work online due to video latency being even worse. Metronomes don't really work for our music because it doesn't follow a standard BPM, it can and should vary.
We did manage a few sessions using JamKazam (https://jamkazam.com/) which works well enough with the tweaks listed above. It's largely restricted to people who know what they're doing, sadly.
This is not a technology bottleneck, it is an implementation bottleneck. Zoom has benefited from the WFH and remote learning movement, but the market is ripe for disruption with a higher quality experience.
Then prior to screen sharing or audio sharing there is a button on the top left for "original audio".
Just turning this off would make a big difference to online music lessons I think. The "speed it up to catch up" technique is fine for speech .. which is the dominant usecase.
1. All contributing musicians broadcast they're ready to "jam"
2. Each musician taps a key on their keyboard (spacebar maybe) when instructed to on the 1st beat of a bar from a metronome written into the software
3. The software works out the delay, and plays back the rest of the audio stream with delay applied
Perhaps that was a user error, but in a craft where getting your frequencies right across the whole spectrum is paramount any kind of lossy optimization can lead to undesirable results.
I wonder if there are parts of Zoom's audio transmission that optimize for the frequency range of the human voice, and everything else gets mangled a little, along the lines of chroma subsampling optimization in the visual world.
As for configuring the settings, it can be easy enough to walk a student through the desktop config but last I checked the tablet config is ridiculous. And even required uninstalling and reinstalling. Would be so happy if that’s no longer the case?
The best part of Zoom imo is being able to share your computer’s audio or the audio from a certain window. That combined with Google Shared Piano is pretty clutch for me.
[edit] oh yeah one other offering to the Zoom gods. I also do a lot of screen sharing, like most of the lesson, and I hate how the video feeds default to a tiny format that can not be resized. A lot of times I need to be able to share my screen AND see my students hands but that’s just impossible now. I could do it with Google Hangouts no problem...
Possibly a bigger issue is having a place where they can work. If you've ever heard a podcast ruined by background noise, you know what I mean.
I think that musicians are particularly adept at mentally filtering out shortcomings of the audio "stack" when they need to, and hear what they need to hear. There are still seminal recordings that are only available as scratchy phonograph records and tapes made with primitive microphones. Maybe it's because we process music at a symbolic level. And it's possible to focus the lessons on the things that can be done remotely. By the time they reach college, they're already playing at an exceptionally high level.
A friend of mine who's a music professor at another university says that he encourages students to use "whatever works" including their cell phones, so that the mechanics of recording don't distract from learning music. They've always had to deal with issues such as students who can't afford professional quality instruments.
Of course they're missing out on ensemble playing.
With the violin, that place is probably above the fiddle, so you have to find a way to dangle your phone, or put a mic on a stand. Especially since you're probably standing to play, for reasons of good posture.
And you want the mic to be close, basically to drown out the effects of typically bad room acoustics unless you're lucky to have a nice recording room. Plus, noise from HVAC and other sources.
And... mainstream recording apps for primarily voice use have built in compression and possibly other artifacts that you have to figure out how to turn off.
And... if you want audio and video, it further constrains your options. You often need your teacher to see you play, to comment on posture and technique issues.
Once it's all working, then the mic in a cell phone and its audio input hardware are actually surprisingly decent. The tiny little condenser mic elements are fairly high fidelity, and an audio codec is pretty much a slam dunk nowadays.
Personally I don't like the sound of a close-mic'd acoustic guitar. Unlike cello and violin, there can be a lot of fret noise and wound strings can be quite squeaky. Classical guitars have different, quieter wound strings under less tension, and nylon for unwound. This is also to reduce fret noise and squeakiness, but personally I prefer the full sound of steel strings. I've been working a lot on getting my fretting technique more clean to eliminate all of that mess and sound more professional, but to be honest I actually prefer to hear my own guitar live in a quiet, resonant room and not through a recording. Bathrooms with tile are great, if a little small! Of course that sounds like complete garbage if you try to record it. But it is absolutely heavenly live :-)
For the doublebass I can imagine you'd be best off with a mic stand at a couple feet away and seeing if you can get a mic that has a very narrow response angle (almost a directed mic).
Cheers and happy recording.
When it was in person, you showed up, played, and were done, good or bad. As a performing musician, I greatly prefer to make a mistake in front of an audience than play perfectly into a microphone.
I'm rehearing with a group of highly amateur musicians, and getting some easily configurable improvement on this would be something we would definitely do, but I missed what the changed settings should be. Thx
Edit: And another problem that's probably even more important for this use case, most common ham radio equipment is simplex (only one person can transmit at a time). In the VHF/UHF range you might be able to engineer a high bandwidth duplex (or TDM) system, but then your range would be limited to a few miles.
> “The biggest change,” Anderson said, “was to run the Opus codec at a higher bit rate.”
I would argue that over the past decade, most humans have been exposed to this kind of processing/compression and our "tastes" reflect our overwhelming exposure to this sound. Just look at how headphones are marketed these days, or any sound system for that matter. "How much bass can I get? Will it knock my socks off?"
The second portion of the video clearly has much for detail and clarity in the higher frequencies, and as a result, can sound a bit "tinnier". It doesn't come across to me as quality at which I'd love to listen to music, but it certainly helps convey the nuances of the harmonies in the piano much, much better.
Advanced Audio:
This option in Advanced Audio will enhance “Original Sound” mode,
allowing you to disable echo cancellation and post-processing
and get rid of compression.
...this setting will also raise audio codec quality from 22kHz to 48kHz, 96Kbps mono/192Kbps stere