I think there a valuble lesson here. Most often when you've gone to hear a demo of Studio Monitors like Yamaha HSs or Adam Audio Monitors, the one thing to always remember. Is that the demo place may not have it setup correctly, and the audio interface and their wiring could be stuffed [the salesperson may not be educated]. One of the brilliant things of Yamaha and Adam Monitors is they are HONEST in what is going on in the electrical and impedence issues (electrical interferrences/conflicts) within your hardware usually. Be patient and not just thinking its a bad manufacting issues of the monitor because it mostly isnt.
For me it's actually the screen itself generating stuff right into the speakers. I'll need to try a different screen to see if changing it could remediate the issue
The issue isnt that the computer was pushing out noise floor. Computer's are natively noisy from all the associated components but its the cheapo behringer interface, almost all modern interfaces outside of beringher are decent nowadays. The internal design in integration & isolation of the DAC prob just got played out and into the speaker. The interface is a soundcard, so all audio would already be already isolated from the computer to the external DAC (which is the interface) so the next logical part of the chain would be where the problem lies.
gotta have good converters and gotta have a good power conditioner for everything to really hear what’s going on also with these speakers a sub won’t hurt 🥴 but the HS7s don’t need a sub so if you have the money find a good used pair of those
@charlieHuntsAudio , my boy you could of got a "Power Conditioner " and that would of eliminated the Noise Reduction on your Monitors , I have some Yamaha HS7 and I had that same problem like you but ever since I bought a Furman M-8X2 it became a game changer sorry but you could of waste less money to fix this issue you had before ..PEACE !
I have a very powerful gamer PC current high-end 300+ watts gpus are radiating insane noise I fixed my setup while keeping analog audio with a Behringer HD400 it seems to be hardly available anymore and you will have to spend money in cables and adapters but it works great not a hiss in my studio monitors
I had this switching between some macbooks and an imac. The imac would put a lot of noise into my audio interface into my speakers. Like through the USB, into the interface and into the speakers! But the macbooks would not.
I think he was using the headphones out from the PC, which went to the Behringer over an analogue cable. That's how the noise from computer components usually ends up making it to speakers. The small mixing desk added even more extra noise on top. His Mac was quieter because the internal headphone outs in Macs are much better designed than the ones you generally find in PCs. However, ultimately, a separate audio interface without a mixer is the best solution, like what he ultimately decided to use.
@@charliehuntsaudio Nah, I just have a little home studio myself and have learned a couple of things over time. The UA interfaces are neat. I'm glad to hear that, mate.
You asked for it: yeah bro that was dumb, it was SUPER obvious what the issue was! You gotta run an external DAC for good sound from a computer! If you're running high gain/high volume and/or have very sensitive speakers, even running an external DAC off USB from your computer can generate similar noises, depending on the DAC, in which case you could either run optical/coax, or use a device that takes the USB input from your computer and splits the power from the data, only passing the data through, then that device gets a separate power input, and it outputs the data from your computer with the cleaner power that's input into it into your DAC. So if you're running USB to your new DAC/mixer thing and you still have a LITTLE bit of noise, that could be your ultimate solution. Check this by comparing the noise floor with your DAC plugged into your speakers with their volume cranked, vs having them on with the volume cranked but nothing plugged into their input. With clean power your DAC should put out basically no noise at all. You could get a much better DAC than that for not a lot of money too but maybe you need the other features / inputs that thing has that a normal DAC doesn't have. But you could also run your old Behringer or any other mixing device for your inputs and just use the DAC for audio out from the computer to the speakers.
If you fom the hifi world you are most certainly correct. If you are running expensive daw software the least you can do is get a decent audio interface, internal or external. I think his mistake doing a professional equipment review on none professional equipment. He has seen the error of his ways. Relying on an economical beringer mixer with known noise and isolation issues with the poorest of converters is not best. Even modist professional internal audio interfaces are always relatively quiet even on the most problematic pc's. The issue using a domestic or hifi audio output method for DAW causes high latency especially when the in and out device is not one unit. That said elaborating on your hifi method a cheap way of good dac is to go optical from your PC at upsampled 96khz to you AV receiver so the AVa receiver is doing the conversion. Force your PC to run 96khz in "sound" preferences. 44.1 or 48khz is a poor choice for this method. This way you avoid the jitter from the PC digital out so your AV receiver will convert at the quality of the built in amps, go 96khz digital as far as possible in the chain. No external dac expense. This option will go quite far but not as good as 4 times over sampling from a high end dac into suitable analogue stereo only system. Use MusicBee player to upsample cd wav and flack files when not using streaming. Use wasapi where available. In PC always playback at 96khz where possible if no dac is over sampling the output.
@@julesc8054 yes I'm coming from a hifi perspective. I agree with half of what you said, and the pro side / audio interface parts. A/v receiver dacs are generally not great (even on fairly high end ones) compared to dedicated dacs. Yep I run 96khz from the computer, and you gotta run in 24 or 32 bit mode for noise floor and digital volume attenuation reasons too. I run straight power amps, receivers suck, and I only use digital volume attenuation the amps have no volume. Very low noise and I have horns. I've used optical to USB, but it's better to use USB with a data/power separating device, in small part because it has more bandwidth and thus you can run higher sampling rates (192khz if you get a USB 3 one, mine isn't so it's limited to 96khz... likely doesn't matter at all) and bit depth, but mainly because your PC is not going to have the same driver control over the DAC with an optical converter, so any special software control they make for it won't work, and some dacs like mine don't have optical in and require USB or maybe USB and coax like mine (coax has similar limitations to optical, maybe worse). High sample rates can help because they improve the characteristics/side effects of the roll off filter in use before the Nyquist frequency, i don't think it has any meaningful benefit in terms of jitter, that'll be based on the DAC, and if the implementation is crap then the computer's output also. I'm not really a subjectivist audiophile guy, l like the ASR objective dac data. Jitter is entirely a non issue on any good modern DAC, there's nothing audible or problematic there.
@@Artcore103 It sounds like an awesome setup. relaxing… some of the older recording before the loudness war. I should apologise I should have said mid-fi. I did mention that a DAC with 4 times oversampling into an analogue system (hi-fi) was a better solution. Your DAC manufacturer want to eliminate the operating system form the audio as much as possible. For Audiophile it’s the best way to go. Out of interest the limitation of the sample rate of 192khz is not the limit of the USB2 standard. RME-audio and others for years, since the late 2000’s have use USB 2 for their interfaces that run 24ch 24bit 192khz + midi @ 6ms latency. Nearly all high channel count audio interfaces of say 64ch to 256ch are PCI internal sound cards with external converters. This is a little less now as more manufacturers use different connectivity options like thunderbolt and usb3 for use with none pci-e commuters. I’m not sure what’s the limitation with your sound card though, maybe the over sampling headroom. What the Nyquist theorem does not specify is the timing of frequency’s, particularly the harmonics, it specifies they can all be created but does not state the timing of the harmonics will be in the original phase relation to when it was created. Remember all the samples affect all the frequencies if they are needed is another debate. For the most part, it makes little difference except on complex music with many harmonics, the sound sage falls apart a little less. As for the jitter, the oversampling and better clocks improves the jitter. Yes, jitter although not an issue is still being improved. RME’s new Steady Clock FS is now infemto seconds (fs). Most clocks are still around 1-2 ps. Remember when CD first came out DAC specs was the thing to look for. DAC’s originally got their name in Audiophile when CD players came in two pieces, the transport and the DAC connected by 110ohm spdif cable. It gave the option of upgrading your CD player without replacing it. These DACs were pure DAC no audio interface. CD Player transports were heavy and a substantial investment. At the time high end computer audio didn’t exist in the mid 80’s, in the 90’s it took 2hrs to convert a single 3min song in Wave lab 2 from wav to mp3. So the DAC’s didn’t have audio interfaces built into them yet so when the steaming era started early 2000’s the need for computers to interface with the then pure DAC was important. The name didn’t change but the functionality did. Now the DAC is more like a preamp with built in audio interface and DAC. Outside of Audiophile it’s called what it is, just a sound card. In studio its an interface, an interface may or may not have da's or ad's those are often referred to separately and connected by adat, dante, madi etc
@@julesc8054 I don't know of any good modern DAC that measures jitter in ns and not low single digit picoseconds, look at the compiled test data by audio science review, he goes into plenty of depth on jitter performance. I'm not saying every DAC is stellar, just that there are dozens that are stellar, and it doesn't correlate with price. I'm not sure what you mean about a DAC going into an "analogue" system... Obviously everything is analogue after the DAC. I mean we could get nuanced regarding class d / class t, but ultimately it's all analogue after the DAC... Digital to ANALOG converter. There are no inherent frequency dependent phase issues with a dacs output. We can test the step response with a 20khz tone and a 100hz tone to see that. This is some hifi DAC black magic marketing bs. The phase coherence is going to be UTTERLY dominated by your speakers, their crossovers, and then we could look at minimum/linear phase EQ effects, since everyone should be using EQ since rooms and speakers are not perfect plus we all have preferences. Technically the roll off filter, depending on it's slope and how into the audio band it goes, could affect phase very slightly perhaps (in the same way eq does), but this is of purely theoretical interest and not audible, especially since cuts are less "problematic" than boosts on paper. If you're into perfectly time aligned drivers, or DSP time alignment, excellent, that can make a difference indeed, specifically off axis and at the crossover points. In terms of absolute phase, we don't hear it, this is about the blending of drivers at the crossover point. After all, even HiFi snobs like eq or tone controls sometimes, bye bye "perfect phase"... But this is largely irrelevant. Even if from 10khz to 20khz the phase was shifted 180 degrees, gradually, you're talking 1/20,000 to 1/40,000 of a second. If this is a smooth transition, which it would be, this is inaudible, as this time period is infinitesimal compared to the lower wavelengths or the perception of music and "timing". Phase issues are about cancellations or peaks created, not about an absolute sense of timing, that's not what we hear. This is why non time aligned speakers and crossovers, and EQd systems, can all sound pretty dang good if they're set up for on axis response at a listening position, but their off axis performance will suffer and so their ultimate fidelity will suffer to the degree that room reflections are part of the sound signature. I appreciate the discussion. Regarding USB 2, I suspect you're quite right, USB isn't so limited as to not be able to support 192khz. Instead, it is simply my particular device that has that limit, and maybe it's some version of USB 1 or 2, idk, I just know if I plug the DAC directly into the computer it supports 192khz, but with this data/power separator in line it's limited to 96khz. I have seen models that support higher sampling rates. And virtually all dacs do or support internal oversampling.
@@Artcore103 You clearly love listening to music and the technology of listening to it. Why not go hire some mics and record a pianist or something in your community. You seem tech savy, have a computer, ears, and maybe inspiration. You could start recording a piano or string quartet at the local music school into a field recorder. Do a quick master in something free and simple like Audition or garage band to start and maybe later record a band in some thing professional and economical like Reaper. Most equipment and mics can be hired for the day. I think you may have a better appreciation of the process challenges and nuances the most avid listeners miss. If you have the ears and the few basic resources for it why not do something rewarding in your community.
HS5's were NEVER made to be your main monitors. They are a modern version of the Yamaha-NS-10's which were used as a pre-release reference monitors to see how it would sound on shitty consumer grade audio tech of the 80's and 90's. Monitor room placement, acoustic treatment and acoustic calibration are essential for producing quality music. Something like the KRK VXT 8's are the minimum I would recommend as main monitors. Pair that with good planer magnetic headphones like the Hifiman HE-400 and THEN something like HS-5's and NS10's for consumer testing even though that is not really relevant anymore as consumer audio quality is much higher. Better option is to test the mix on car stereo, apple earpods and samsung earpods and phone speakers and you are covered.
HS-5 is definitely not a modern version of the NS-10’s. The frequency response is different, the low-end on NS-10’s rolls off at 100Hz, while the HS-5 rolls off at 60 Hz.
Add the sub to them and only have them doing mids and highs and I bet you retract that statement… I thought the same as you at one point before I heard them with a sub. I was wrong. Btw I use ATC’s in the professional setting and I’m saying this… so.
@@busywl69 yea I feel you. They are more talented than myself. I need to give myself an advantage so I use analog on the way in as well for my mix bus. For me personally I’m not able to get plugins to sound even remotely as good as hardware. I know some swear by it but I def prefer analog to my ears with my workflow.
Looool bully, nah seriously they ate a fantastic starting point it just depends on the use scenario, my main use is orchestral soundtracks and for string sections they are great but for the rest I now switch over to my barefoots
![[FULL EP.73] เปิดบ้าน 70 ล้าน "แป้ง Zbing" ยกมาทั้งญี่ปุ่น l เคหสถานบานปลาย l One Playground](http://i.ytimg.com/vi/YBnJhTt07Pg/mqdefault.jpg)
I think there a valuble lesson here. Most often when you've gone to hear a demo of Studio Monitors like Yamaha HSs or Adam Audio Monitors, the one thing to always remember. Is that the demo place may not have it setup correctly, and the audio interface and their wiring could be stuffed [the salesperson may not be educated]. One of the brilliant things of Yamaha and Adam Monitors is they are HONEST in what is going on in the electrical and impedence issues (electrical interferrences/conflicts) within your hardware usually. Be patient and not just thinking its a bad manufacting issues of the monitor because it mostly isnt.
Good on you for making a correction video, dude! I'm glad your new interface is allowing you to have a better experience with your monitors.
Thanks brother appreciate the positive vibes 🙂
For me it's actually the screen itself generating stuff right into the speakers. I'll need to try a different screen to see if changing it could remediate the issue
Great information and video, thanks for posting ✨🙌
The issue isnt that the computer was pushing out noise floor. Computer's are natively noisy from all the associated components but its the cheapo behringer interface, almost all modern interfaces outside of beringher are decent nowadays. The internal design in integration & isolation of the DAC prob just got played out and into the speaker. The interface is a soundcard, so all audio would already be already isolated from the computer to the external DAC (which is the interface) so the next logical part of the chain would be where the problem lies.
Ye I definitely realised that 😅 lol by cheap buy twice but to be honest I think it was just an old aged and tires unit
Love my HS5. I’m glad you get to enjoy them too.
gotta have good converters and gotta have a good power conditioner for everything to really hear what’s going on
also with these speakers a sub won’t hurt 🥴 but the HS7s don’t need a sub so if you have the money find a good used pair of those
@charlieHuntsAudio , my boy you could of got a "Power Conditioner " and that would of eliminated the Noise Reduction on your Monitors , I have some Yamaha HS7 and I had that same problem like you but ever since I bought a Furman M-8X2 it became a game changer sorry but you could of waste less money to fix this issue you had before ..PEACE !
One of these days< I'll dig out my old Yamaha S55, buy an amp and set them up.
I have a very powerful gamer PC current high-end 300+ watts gpus are radiating insane noise I fixed my setup while keeping analog audio with a Behringer HD400 it seems to be hardly available anymore and you will have to spend money in cables and adapters but it works great not a hiss in my studio monitors
I had this switching between some macbooks and an imac. The imac would put a lot of noise into my audio interface into my speakers. Like through the USB, into the interface and into the speakers! But the macbooks would not.
Wow I watched solely because you retracted you barely ever see any retracted vloggies that’s are this real good wowed impressed 🍒🍒🍒🍒✨✨✨💥🏆🏆🏆🏆🏆🥂🥂🥂🥂🍾🍾🍾
Thank you I appreciate you taking the time to send me that
Using that Behringer mixer ... but what was your interface?
I think it was the behringer mixer.
I think he was using the headphones out from the PC, which went to the Behringer over an analogue cable. That's how the noise from computer components usually ends up making it to speakers. The small mixing desk added even more extra noise on top.
His Mac was quieter because the internal headphone outs in Macs are much better designed than the ones you generally find in PCs.
However, ultimately, a separate audio interface without a mixer is the best solution, like what he ultimately decided to use.
Awesomely put! And u had my Setup absolutely correct, u aren't my electrician are u loool. Thanks for the concise comment
I'm using a universal audio interface now
@@charliehuntsaudio Nah, I just have a little home studio myself and have learned a couple of things over time. The UA interfaces are neat. I'm glad to hear that, mate.
You asked for it: yeah bro that was dumb, it was SUPER obvious what the issue was! You gotta run an external DAC for good sound from a computer! If you're running high gain/high volume and/or have very sensitive speakers, even running an external DAC off USB from your computer can generate similar noises, depending on the DAC, in which case you could either run optical/coax, or use a device that takes the USB input from your computer and splits the power from the data, only passing the data through, then that device gets a separate power input, and it outputs the data from your computer with the cleaner power that's input into it into your DAC. So if you're running USB to your new DAC/mixer thing and you still have a LITTLE bit of noise, that could be your ultimate solution. Check this by comparing the noise floor with your DAC plugged into your speakers with their volume cranked, vs having them on with the volume cranked but nothing plugged into their input. With clean power your DAC should put out basically no noise at all. You could get a much better DAC than that for not a lot of money too but maybe you need the other features / inputs that thing has that a normal DAC doesn't have. But you could also run your old Behringer or any other mixing device for your inputs and just use the DAC for audio out from the computer to the speakers.
If you fom the hifi world you are most certainly correct. If you are running expensive daw software the least you can do is get a decent audio interface, internal or external.
I think his mistake doing a professional equipment review on none professional equipment. He has seen the error of his ways. Relying on an economical beringer mixer with known noise and isolation issues with the poorest of converters is not best.
Even modist professional internal audio interfaces are always relatively quiet even on the most problematic pc's. The issue using a domestic or hifi audio output method for DAW causes high latency especially when the in and out device is not one unit.
That said elaborating on your hifi method a cheap way of good dac is to go optical from your PC at upsampled 96khz to you AV receiver so the AVa receiver is doing the conversion. Force your PC to run 96khz in "sound" preferences. 44.1 or 48khz is a poor choice for this method. This way you avoid the jitter from the PC digital out so your AV receiver will convert at the quality of the built in amps, go 96khz digital as far as possible in the chain. No external dac expense.
This option will go quite far but not as good as 4 times over sampling from a high end dac into suitable analogue stereo only system.
Use MusicBee player to upsample cd wav and flack files when not using streaming.
Use wasapi where available. In PC always playback at 96khz where possible if no dac is over sampling the output.
@@julesc8054 yes I'm coming from a hifi perspective. I agree with half of what you said, and the pro side / audio interface parts. A/v receiver dacs are generally not great (even on fairly high end ones) compared to dedicated dacs. Yep I run 96khz from the computer, and you gotta run in 24 or 32 bit mode for noise floor and digital volume attenuation reasons too. I run straight power amps, receivers suck, and I only use digital volume attenuation the amps have no volume. Very low noise and I have horns.
I've used optical to USB, but it's better to use USB with a data/power separating device, in small part because it has more bandwidth and thus you can run higher sampling rates (192khz if you get a USB 3 one, mine isn't so it's limited to 96khz... likely doesn't matter at all) and bit depth, but mainly because your PC is not going to have the same driver control over the DAC with an optical converter, so any special software control they make for it won't work, and some dacs like mine don't have optical in and require USB or maybe USB and coax like mine (coax has similar limitations to optical, maybe worse).
High sample rates can help because they improve the characteristics/side effects of the roll off filter in use before the Nyquist frequency, i don't think it has any meaningful benefit in terms of jitter, that'll be based on the DAC, and if the implementation is crap then the computer's output also. I'm not really a subjectivist audiophile guy, l like the ASR objective dac data. Jitter is entirely a non issue on any good modern DAC, there's nothing audible or problematic there.
@@Artcore103
It sounds like an awesome setup. relaxing… some of the older recording before the loudness war.
I should apologise I should have said mid-fi. I did mention that a DAC with 4 times oversampling into an analogue system (hi-fi) was a better solution.
Your DAC manufacturer want to eliminate the operating system form the audio as much as possible. For Audiophile it’s the best way to go.
Out of interest the limitation of the sample rate of 192khz is not the limit of the USB2 standard. RME-audio and others for years, since the late 2000’s have use USB 2 for their interfaces that run 24ch 24bit 192khz + midi @ 6ms latency. Nearly all high channel count audio interfaces of say 64ch to 256ch are PCI internal sound cards with external converters. This is a little less now as more manufacturers use different connectivity options like thunderbolt and usb3 for use with none pci-e commuters. I’m not sure what’s the limitation with your sound card though, maybe the over sampling headroom.
What the Nyquist theorem does not specify is the timing of frequency’s, particularly the harmonics, it specifies they can all be created but does not state the timing of the harmonics will be in the original phase relation to when it was created. Remember all the samples affect all the frequencies if they are needed is another debate. For the most part, it makes little difference except on complex music with many harmonics, the sound sage falls apart a little less.
As for the jitter, the oversampling and better clocks improves the jitter. Yes, jitter although not an issue is still being improved. RME’s new Steady Clock FS is now infemto seconds (fs). Most clocks are still around 1-2 ps.
Remember when CD first came out DAC specs was the thing to look for.
DAC’s originally got their name in Audiophile when CD players came in two pieces, the transport and the DAC connected by 110ohm spdif cable. It gave the option of upgrading your CD player without replacing it. These DACs were pure DAC no audio interface. CD Player transports were heavy and a substantial investment. At the time high end computer audio didn’t exist in the mid 80’s, in the 90’s it took 2hrs to convert a single 3min song in Wave lab 2 from wav to mp3. So the DAC’s didn’t have audio interfaces built into them yet so when the steaming era started early 2000’s the need for computers to interface with the then pure DAC was important. The name didn’t change but the functionality did. Now the DAC is more like a preamp with built in audio interface and DAC. Outside of Audiophile it’s called what it is, just a sound card. In studio its an interface, an interface may or may not have da's or ad's those are often referred to separately and connected by adat, dante, madi etc
@@julesc8054 I don't know of any good modern DAC that measures jitter in ns and not low single digit picoseconds, look at the compiled test data by audio science review, he goes into plenty of depth on jitter performance. I'm not saying every DAC is stellar, just that there are dozens that are stellar, and it doesn't correlate with price.
I'm not sure what you mean about a DAC going into an "analogue" system... Obviously everything is analogue after the DAC. I mean we could get nuanced regarding class d / class t, but ultimately it's all analogue after the DAC... Digital to ANALOG converter.
There are no inherent frequency dependent phase issues with a dacs output. We can test the step response with a 20khz tone and a 100hz tone to see that. This is some hifi DAC black magic marketing bs. The phase coherence is going to be UTTERLY dominated by your speakers, their crossovers, and then we could look at minimum/linear phase EQ effects, since everyone should be using EQ since rooms and speakers are not perfect plus we all have preferences. Technically the roll off filter, depending on it's slope and how into the audio band it goes, could affect phase very slightly perhaps (in the same way eq does), but this is of purely theoretical interest and not audible, especially since cuts are less "problematic" than boosts on paper. If you're into perfectly time aligned drivers, or DSP time alignment, excellent, that can make a difference indeed, specifically off axis and at the crossover points. In terms of absolute phase, we don't hear it, this is about the blending of drivers at the crossover point. After all, even HiFi snobs like eq or tone controls sometimes, bye bye "perfect phase"... But this is largely irrelevant. Even if from 10khz to 20khz the phase was shifted 180 degrees, gradually, you're talking 1/20,000 to 1/40,000 of a second. If this is a smooth transition, which it would be, this is inaudible, as this time period is infinitesimal compared to the lower wavelengths or the perception of music and "timing". Phase issues are about cancellations or peaks created, not about an absolute sense of timing, that's not what we hear. This is why non time aligned speakers and crossovers, and EQd systems, can all sound pretty dang good if they're set up for on axis response at a listening position, but their off axis performance will suffer and so their ultimate fidelity will suffer to the degree that room reflections are part of the sound signature.
I appreciate the discussion.
Regarding USB 2, I suspect you're quite right, USB isn't so limited as to not be able to support 192khz. Instead, it is simply my particular device that has that limit, and maybe it's some version of USB 1 or 2, idk, I just know if I plug the DAC directly into the computer it supports 192khz, but with this data/power separator in line it's limited to 96khz. I have seen models that support higher sampling rates. And virtually all dacs do or support internal oversampling.
@@Artcore103
You clearly love listening to music and the technology of listening to it.
Why not go hire some mics and record a pianist or something in your community. You seem tech savy, have a computer, ears, and maybe inspiration.
You could start recording a piano or string quartet at the local music school into a field recorder. Do a quick master in something free and simple like Audition or garage band to start and maybe later record a band in some thing professional and economical like Reaper. Most equipment and mics can be hired for the day.
I think you may have a better appreciation of the process challenges and nuances the most avid listeners miss.
If you have the ears and the few basic resources for it why not do something rewarding in your community.
Behringer. There’s your problem
🤪😅 u bully
HS5's were NEVER made to be your main monitors. They are a modern version of the Yamaha-NS-10's which were used as a pre-release reference monitors to see how it would sound on shitty consumer grade audio tech of the 80's and 90's. Monitor room placement, acoustic treatment and acoustic calibration are essential for producing quality music. Something like the KRK VXT 8's are the minimum I would recommend as main monitors. Pair that with good planer magnetic headphones like the Hifiman HE-400 and THEN something like HS-5's and NS10's for consumer testing even though that is not really relevant anymore as consumer audio quality is much higher. Better option is to test the mix on car stereo, apple earpods and samsung earpods and phone speakers and you are covered.
lol
Thanks for your input, they only replaced my previous monitors briefly, I'm running a pair of barefoots alongside them now 😀
HS-5 is definitely not a modern version of the NS-10’s. The frequency response is different, the low-end on NS-10’s rolls off at 100Hz, while the HS-5 rolls off at 60 Hz.
Buuuuuuut they still sound like crap 😆
Add the sub to them and only have them doing mids and highs and I bet you retract that statement… I thought the same as you at one point before I heard them with a sub. I was wrong. Btw I use ATC’s in the professional setting and I’m saying this… so.
I hear you, but I've put them up against 12 other monitors in the same price bracket and they lose every single time. @@erickarge1838
@@erickarge1838 Talented producers can make great music with a pair of headphones and KRKs. But snobs gonna snob of course.
@@busywl69 yea I feel you. They are more talented than myself. I need to give myself an advantage so I use analog on the way in as well for my mix bus. For me personally I’m not able to get plugins to sound even remotely as good as hardware. I know some swear by it but I def prefer analog to my ears with my workflow.
Looool bully, nah seriously they ate a fantastic starting point it just depends on the use scenario, my main use is orchestral soundtracks and for string sections they are great but for the rest I now switch over to my barefoots