APZX

Well, that guy isn't "technically" wrong but there was a lot glossed over in that simplified explanation. So, now frequency response is just one specification of an amplifier that you should pay attention and really it is one of the most useless for actually telling you much about it. Beyond telling you what frequency ranges the amplifier will amplify it also implies the phase margin of the amplifier. Essentially, the amplifier has a more phase aligned output throughout the actual audible band in "theory".* As I said before you don't "technically" need additional line outputs to use a headphone amplifier. There are disadvantages to using one set of line outputs, but it isn't necessary. If you're using the built in volume control on your audio interface then you've got two stages where you can introduce either gain or attenuation before you actually hear it. Right or wrong really is a semantics kind of thing on paper. In practice the fewer things in your monitoring signal path the better because there are less things to get in the way of what you're hearing. I know before I bought my current audio interface, RME FireFace UFX+, I considered a lot of audio interfaces, but I also had some different requirements than you did. However, I also don't use the audio interface for monitor control. That is done via an external unit in my case and I just pipe over an unattenuated signal to my monitor controller, which has a banging headphone amp lol. So, the Tascam is generally well liked for what it offers. The Steinberg stuff is always pretty highly liked too. I like Focusrite stuff too so something like the 6i6 or even 18i8 are also quite comparable. But another possibility is Presonus. There are lots of audio interfaces out there. You have to figure out your requirements to see what it is you actually need and may need in the future. I can't tell you what those are. You spell out your requirements and select an audio interface based on that. If for example you want to use Universal Audio plugins then you may actually consider getting something like the Universal Audio Arrow or splurging for an Apollo Twin mkII. I only brought this point up because I find it smarter to spend however much money you need at first to get what you actually need. * - Extra Geek talk. Now, I don't really need to read much into the specifications of the Lake People G103-P because I know what it is built around, a LM1876. That is the datasheet to the device at the heart of the G103-P. Now, this device is designed to drive 4-8 ohm speakers. However, because it is driving such high impedance loads the chip's performance will be much better in several aspects. If you look at the datasheet on page 10 for Figure 10 to Figure 15 you'll see a bunch of graphs that are titled THD + N vs Output Power (THD + N stands for Total Harmonic Distortion plus Noise). Now, I doubt you're going to be running this anywhere near 1W output power. More than likely you'll be running it around 50-75mW. So, you're going to want to pay attention to that first decade on each graph which represents 10mW to 100mW. It is also important to note that these are specified at particular loads and frequencies, so pay attention to what it says in the upper left of each graph. So, it looks like THD + N will probably be around 0.05%-ish, which is quite good actually. There is also Figure 19 on page 12 that specifies the amount of output power into a given load. It only goes 40 ohms, but it looks like it is starting to level out. The maximum output power is probably going to be ~4W. There are other important factors at play here that aren't specified often that you should always look for. These include the Slew Rate of the amplifier, which given that this is a power amplifier is quite high at 12V/us** (this is a worse case figure). One factor that is a bit lacking is the Crosstalk though, at a typical 80dB, but again should be much better because it is very unlikely that you're going to be driving the headphones with a nearly 11Vrms signal. That is just absurd. There are plenty of other things to keep in mind too. Figure 29 is a good one for showing how the phase of the amplifier varies, Figure 28 is also helpful for showing how good the amplifier is at rejecting common mode noise vs frequency. Like from this datasheet you can pretty much guess exactly how good this amplifier will be in terms of specifications This is why I say specifications are important. Now, they could improve the CMRR of the LM1876 by using an external opamp and running the output of that into the LM1876's non-inverting input, but I can't really find much more on the G103-P other than it uses a LM1876. For further reading to better understand these specifications I highly suggest reading Rane's Note on Audio Specifications. So, why do I say the guy is "technically" correct? Well, because like most things in life there is a grain of truth to what he is saying, but it isn't the whole truth. See, by his logic then by extension higher impedance speakers need more powerful amplifiers, which isn't actually the case. See any power amplifier is really just trying to swing the output voltage to the voltage rails of the power supply. That is to say if you're running a 60V power rail then it can produce a voltage difference of 60V across the load, aka the speaker. So, then you may quite rightly ask, "Then why do power amplifiers need these massive heatsinks if all a power amplifier is doing is changing voltage?" Well, this is where Ohm's Law comes into play. In a nutshell Ohm's Law says that if you have resistance and there is a voltage difference applied across that resistance then current must flow as a consequence. See, you know the resistance of the speaker, which say for the sake of simplicity and the like is 8 ohms, then that 60V across that 8 ohms requires a certain amount of current to flow. According to Ohm's Law, the Current (I) is equal to the Voltage (V) divided by the Resistance (R) or I = V/R, I = 60V/8 ohms = I = 7.5 amps. That is why an amplifier needs all that heatsinking, because it has to be able to source A LOT of current as a consequence of Ohm's Law. Now, the main issue with many power amplifier is that they actually become current limited. This means that in order to sustain that 60V across the load the power amplifier can no longer source enough current to maintain the relationship between Voltage and Current as required by Ohm's Law. Headphones must obey this as well, and why I originally said that the problem with most headphone outs is that the output device is simply inadequate to drive the desired load. Looking at a common audio opamp, such as the NE5532, there is a specification called Output short-circuit current labeled as Ios. This spells out the ABSOLUTE maximum amount of current that the NE5532 can deliver. There are three figures listed for it, which are "Min", "Typ", and "Max". Assuming the "Typ" or Typical value of 38mA along with the Maximum peak-to-peak output-voltage swing (pretty self explanatory, but says the maximum voltage difference the opamp can produce) you can actually figure out the maximum amount of resistance that the NE5532 can drive using Ohm's Law. In this case R=V/I = R= 24V/0.038A = 631 ohms. Further, since we know both the voltage and the current, we can also estimate the power, which is about 912mW (basic estimates are easy as P = IxE, where E is actually Voltage but is based on slightly different terminology). So, for your DT 880s, a NE5532 could drive them quite well The problem is most headphone outs don't use a NE5532 because while it is a cheap opamp, it costs more than a NJM4558, which is often used. In 10,000 quantity the NE5532 is about $0.29 vs the NJM4558 at $0.10. Why spend an extra 19 cents, when you don't have to when the output is just "adequate"? The simple answer is they don't. ** - Super duper extra Geek Talk Slew Rate is an interesting specification. It is basically how fast the amplifier can change the voltage or current in certain situations. So, for the LM1876 that is 12 volts per microsecond or V/us or for fancy folks V/μs. This means that if the LM1876 is at say 0V and all of a sudden it needs to jump to +12V then it will take it 1 us to do that. Now, say the LM1876 needs to move to -12V then it will take 2 us for it to move to that. 1 us to move to 0V and 1 us to move to -12V. This specification fundamentally limits the maximum frequency that amplifier can reproduce in a linear fashion. Assuming that the LM1876 is running on +/-15V (a common value for audio circuits and I'd be surprised if the G103-P is running much higher or lower than this) then that means the maximum peak voltage the LM1876 needs to be able to swing is 30Vpk. So, if it is quoted to be able to reproduce 150,000Hz then to do so in a linear fashion requires that the LM1876 have a Slew Rate of 28.278V/us. Now, the LM1876 could very be able to operate at that kind of slew rate depending on the load that it is driving. I'm just using 12V/us as a worst case scenario kind of thing. There also isn't a graph to show how the Slew Rate changes vs anything so I couldn't even hazard a guess at how it is performing. However, that worst case figure gives an upper bound of frequency right around 63,700Hz anyway. But it still pails in comparison to a lot of other parts. The NE5532 for example is about 9V/us, the TL07X series have 13V/us, and high performance opamps like the OPAx134 series are 20V/us or OPA161x is 27V/us. Also, super high slew rates aren't necessarily a good thing either as they can lead to opamp instability and may require frequency compensation to stabilize the opamp, typically in the form of a compensation capacitor. A classic example of this is with the NE5534 which requires a compensation capacitor whenever the gain is below three, typically a 22pF. And even then I'm not going to enter into the topic of Gain-bandwidth product or GBW and how it is also a factor to consider with amplifier stability and how all of this comes together to help explain whether an amplifier is actually stable. Definitely, a lot of good reading on the topic if you want to explore more into it though.

It would totally work! If you're finding your DT 880s too quiet or perhaps maybe lacking then a proper dedicated headphone amp will be the ticket. Likewise with your MDR-7506s and moving to the Roland. With that said, if you're happy with the headphone output on the UR22 and you need to drive monitors as well then you can. However, it requires that you have the ability to "split" the line output signal. This can be done with something like this, and if you're handy with a soldering iron you can make them yourself The reason this works is because in transferring audio the thing of concern is the AC Voltage. So, as I mentioned in my first post audio systems operates on Impedance Bridging. The fundamental principle in action here is that your audio interface will have a very low output impedance, on the order of a few ohms or less, and the input to any given device will typically be on the order of at least 10 kilohms. This means that very little current actually flows into the receiving device and you're effectively measuring the voltage. Then since the devices receiving the AC Voltage are connected in parallel they will have the same voltage potential across them. It is actually quite an elegant solution honestly. Your upgrade plan has nothing "technically" wrong with it all. It is very much equipped for the intended purpose of giving you two headphone outputs with one of those outputs having tons more power than you do you now. Whether it sounds "better" is something only you can decide. If I may make one suggestion. Think about what you're going to need in the future in terms of your audio interface. If you want more ins and outs then buy it now. You may have plans in the future for an outboard synth or an outboard buss compressor and those will eat up valuable analog ins and outs. I'm not saying that the Rubix 24 is bad or the like, but it is a small interface without much actual I/O digital or analog. You may be working 100% ITB with maybe a need for a mic occasionally. For that it is perfectly equipped. I know personally I'd be too limited by it immediately. So, just think about what you actually want to do in the future and buy based off that I've got a fairly complicated setup so I've had to learn to deal with the sorts of issues that pop up. I've had to deal with ground loops, bad cables, not enough I/O, varying power voltages, etc . . .. This can be a complicated mess to sort out. So, if you ask me a question about the "technical details" I can probably explain it. However, when you start getting into subjective things like will it sound better or the like, well only really you can decide that. I mean there are some obvious things like if you want more detail in the bass region then you've got to do the whole room treatment things and get bigger/better speakers. There really is not an easy way around that sort of thing. However, if you want to know why your square waves don't look like square waves on an oscilloscope in the real world or connecting things up. I've got you covered.

Short concise answers first, and then I'll follow up with technical geeky mumbo jumbo. 1 - Technically, a line output from your audio interface should be able to drive many line inputs because they use Impedance Bridging, which relies on the core assumption that both the line output and line input are done properly. Though to do this does require have cables or a box that allows you to split the signal. However, if you have multiple outputs then it affords you more flexibility in how you can set things up. 2 - I don't see any reason why not. Just well made cables I suppose. You know for cable longevity and reliability. All right time for the technical geeky mumbo jumbo if you want some further explanation on things. Lets talk about that headphone output on the Roland Rubix 24, which Roland actually gives a maximum power specification (well at least sorta kinda, but better than nothing). Anyway, so a headphone output is really just a mini power amplifier because at the end of the day all a pair of dynamic headphones (like the two you listed) are just tiny speakers. So, Roland specifies the output power as 20mW into 47 ohms. You can estimate the amount of SPL your headphones will have based on this. It isn't perfect and you could sit down and try and do fancy mathematics to get at a more concise answer, but this is just quick and dirty. However, simple cross multiplication should "ballpark it". So, for the Sony MDR-7506 headphones you can expect that about 27mW maximum will be on tap. This is calculated by the classic (20mW/47 ohms) = (X/63 ohms). From here you get (63 ohms * 20mW) / 47 ohms. The ohms cancel out and you're left with 26.8mW. At any rate the MDR-7506's have a specified sensitivity of 106dB/W/m. This would give a peak SPL in excess of 120dB SPL. So, I'd say there is probably plenty of room there. Next up are those Beyerdynamic DT 880 250 ohm headphones. So, these are rated at 96dB/W/m (not stated like that, but at any rate). So, doing the same trick as before the Roland unit should produce around 106mW into the DT 880s. This gives a peak SPL around 116-117dB. Now, these numbers are estimations on the maximum SPL that you can expect from the Roland and can be eaten up quite quickly by very big and powerful transients. So, unless you want more volume I've never really understood the point of headphone amps 99% of the time personally, but I digress. Most of the time headphone outputs simply use an inadequate device for the output stage without enough bulk energy storage (big capacitors on the power rails). My best guess is that the Roland is probably using an "okay" opamp as the output stage. I'd be willing to bet that the headphone output on the Roland is either a single JRC4558 or two JRC4558s that are paralleled up. I doubt that they're using anything better, and for a whole mess of reasons this part is basically barely & adequately passable part for the job, but again I digress. This then I guess begs the question why would I say much about that? Well see the output is probably just an opamp anyway. This means that it'd have no problems driving any line input. It won't be balanced and it will probably be noisier than a dedicated line output given that it is being used as a power amp, BUT there is nothing "technically" wrong with using it that way. Not best practice of course. I also feel it necessary to say something about the impedance balanced output. Really the trick here is to understand what "impedance balanced" actually means. In a nuthsell this means that the cold lead or "- lead" is not actually driven by anything, but instead presents the same "impedance" to ground that the hot lead or "+ lead" presents. This will eliminate most things like ground loops and hum, but not much else, which for most things is an acceptable compromise in cost. The other common option is to drive both the hot and cold leads and this provides much better rejection of all sorts of things, but it is more costly. Why use two opamps, four bypass capacitors, and five resistors when you can use one opamp, two bypass capacitors, and six resistors. Takes less board space and costs less overall to do the single opamp. Hopefully, you found something useful in there.

@HoboKa Sorry to say, but you also got the wrong tune on YouTube. It should be

@Dextastic You don't have use the gate as a full on gate either. Just use it to turn down the sustain part automagically

I guess it would be the way that instruments are placed. You can have extremely short reverbs that'd do pretty much what I'm hearing. But you are right there isn't any reverb there. Some tips. The single largest issue you have going on that I can hear is that everything is fighting for the number one spot. That just doesn't happen. For one sound to have something it has to take away something from another. Choose one sound you want to be the focal point. Then make everything fit around that one sound, for example the strings. Now, the strings aren't going to be your instrument here. So, what is another big focus? How about those timpanis (they sound like them or they could be taikos, but the attack is different). There is a lot of sustain on them. You've now got a choice to make. Do you just let their sustained portion run into one another creating a constant rumble? Or do you space hits so that the sustain of each hit ends before the next hit? Or do you try and quell the rumble in the mix? Of course you still have to consider the rest of the percussion. The snare? Where is it going to sit in relation to the timpani and the strings? What about the hats? How are they going to interact with the strings, timpani, and snare? How about that guitar (again I think it is a guitar, almost like a picked bass)? How are you going to integrate that in with the rest of the instrumentation? These aren't just mix focused questions, they are compositional questions first and foremost because the start of a solid mix is where the composition has everything basically where it ought to be. Sure some sounds might need a nudge or two, but in general things should be working in the composition. Well before you ever hit the notion of mixing.

You're more patient an individual than myself lol.

Got my entry up. The one thing I've always admired about the Metroid Prime OST is its ability to be both beautiful and scary at the same time.

I rather like Wave Alchemy's stuff. They have some free stuff, and they have a free EDM oriented kick drum pack, Club Kicks. Give a look to their free samples as well. Some pretty solid stuff there including some quite nice 707 samples (quite an awesome kick honestly), some samples of a Vermona DRM (slightly biased as I have one of these myself, but excellent samples), and MS-20 Mini (these are some pretty darn solid and tight samples).

Frankly, it is kind of a crapshoot. It isn't like in mixing where you can make a very good mix just using the bundled with the DAW. However, with synths or samples you've unfortunately gotta spend $$$. It kind of sucks to say it, but it is the truth. I mean I don't do orchestral music because I find the libraries 1 - Too Big 2 - Too Expensive and 3 - Too complicated. And this is coming from a guy who loves modular synths with more knobs than you can throw a stick at. But that doesn't mean everyone should be all synths all the time. I enjoy listening to well crafted orchestral takes on songs, and I like using real strings from time to time myself, but then again I kind of obfuscate them in my tracks because I just want the hint of realism and their additional qualities rather than having them as my main focus. But I do have some quite nice soundfont pianos because there a couple of good ones that are free. The best probably being the Salamander Grand Piano. But I digress. If you recognize their intent then I do feel it is probably best to let them know that with their current tools they're unlikely to achieve what they want or are going for. This isn't really a disservice, but frankly a truth. Sure it is a bit cold, but what are you to do? I mean you don't have to come out and be a total downer when you say it. A lot about how you'd approach the situation is really just how you say it. At least that is what I've found.

If you want something overkill for 99% of the situations you'll encounter then yeah Ozone would be a great choice. However, if you want stuff that'll be more useful then @shadowpsyc brings up an excellent point. Monitoring. You can't really know if you're doing good or harm without something telling you honestly that you are doing good or harm to the sound. So, I +1 that comment. With that being said, I do also understand the want or desire for extra VSTs or such for mixing or effects. I personally really dislike PEQ2 because is in HQ mode it has unreported latency that can cause problems. Though as a main EQ I'd find it too fiddly most of the time because it is a Parametric EQ, and they certainly do have their uses. However, most of the time I just need a couple of simple bells and a couple of shelving filters for my EQ work with HPF and LPF. The EQ I use most of the time when I'm mixing is one modeled after a Neve 1081 (how close it sounds, I don't know, nor do I care). What I care about is how fast I can get what I want out of it. In that regard it is perfect. Limited, but plenty flexible for most situations you'll ever encounter in mixing. I guess my biggest suggestion is to find a set of tools that you like and can use quickly. That is probably the bigger thing here. Ozone is a big giant collection of processing that would probably be useful from time to time, but not often enough I'd say to go and get right now.

I was not directing it at you per se, you just happened to say it Written language is a bit fickle and forums are sometimes kind of hard to clearly state what you want to say without sounding like a twat sometimes.

That there I feel brings into question a larger more overriding notion that is merely being glossed over here. You even mention a good example yourself, Trance versus Jazz. I'm a Trance head for the record. So, repetition in Trance is very much a thing, and is really a core tenet of the genre as a whole. However, if you go listen to Trance you'll find that it is actually less repetitive than you might think. The texture of the track is varied and modulated throughout. Someone who listens for chord changes and progressions is likely to be bored as a result of this. I think Trance is actually kind of a bad choice in this comparison and would instead choose Minimal as it is far more repetitious, or you could even throw Ambient in there. Even then those two genres have their own kind of movement to them that should be taken into consideration. To simply ignore this is missing a core principle of the genre at hand, and not looking at things from the appropriate perspective. Just my $0.02.

@Dextastic I've got nothing against Rock or Metal, heck I love me some Prog or Psychedelic Rock and some Metal stuff. It is just that really most of the time it is the people behind the guitars rather than the guitars themselves that I have an issue with. Not trying to imply anything just a "general observation". Also, no problem giving some techniques for dealing with things in a mix. I've done a few Hard Rock and a couple of Metal Mixes and there is definitely some different things than you might think. Something else to consider when mixing is that tricks that you may have used in one mix may come in handy for a mix that is completely and utterly unrelated in both genre and style than where you learned it. You'd think that I'd never turn to distortion in a non-rock/metal mix, but surprisingly it comes in handy in all sorts of places. In particular I like using it on bass too make it easier the ear to find it in the mix if it is quite dull.

@Dextastic With the compressor pretty much, but you'll need to adjust based on taste and even play with different compressors as they're attack and release curves will change the way the sound behaves. Just gotta experiment a bit. I can tell you that for this particular application I really really like vladg's Molot. I know it is written in the Cyrillic alphabet, but there is an English option and I like to do this in the "Alpha" mode because it isn't quite as aggressive. It is preference and I've used different compressors for the application just based on various things from changes to the timbre to fitting better in with the other instruments. Compression in general is quite a complex topic and it'd take me way too long to explain in any reasonable detail about how it works or the various ways to use to manipulate space or even forcing one without a sidechain input to react in an almost dynamic EQ kinda way. But with that being said you've really just gotta experiment with it to find something that really works for you. I can tell you that I have a lot of compressors because I find unlike EQs I can get more color out of them and that is typically what I end up doing on a lot of sounds. Rather than using them strictly for mixing. I say compression on the guitars because even though distorted guitars are already quite compressed it isn't the end all be all to further coloring them with compression Much to learn about that you do! But some things you can try to give the guitars some more weight. Put on a different hat for distortion for a minute. Think of distortion as a way to generate new and extra harmonics that you can then manipulate to your advantage. For example you could try some frequency selective parallel distortion to add in some extra girth that you can then mix in with the original to give them more heft if they're missing it. Kind of like the old trick of gating a tone generator from a kick to make it beefier in the mix. Since you have 4 guitars in this particular case then I would look at doing something a bit different personally. I'd take two of them and hard pan them and leave them mostly alone (EQ, compression, etc.) but pan the other's about midway left and right then play with distortion on them and make those heavier sounding. This would likely widen the mix with two different guitar layers at two different stereo positions, but would also have the effect of making the center seem "heavier" and leaving the sides "lighter" further adding to width without really sacrificing any mono compatibility As for the synths, I mean they kinda work. The brass is actually a pretty good sounding brass stab, but it sounds like it came from a ROMpler of some description. I like the piano quite a bit, and I love the sync lead but it is so buried and overshadowed by everything else in that section that it kind of loses its impact While this may sound kind of biased or such, I really really dislike guitars lol. Not because there is anything wrong with them, but just because I'm not a guitar guy and everyone talking about them makes it sound like there are humongous differences between ancient humbucklers and modern ones. Look Solid body vs Hollow Body? Easily understandable differences in sound. But I feel arguing about a 139048 year old humbuckler is like the synth guys saying that there is a huge difference in the way a 50 year old Minimoog sounds versus a clone from like Studio Electronics. Look is there a difference? Probably, but is it worth all that money for it? Probably not. I'm a synth head and I love synths. It is easier to impress me with guitars than it is with synths. Suuns' 2020 (warning epileptic seizure warning) has a guitar sound that I love because I've never really heard it before and plus it is super creepy and unsettling. The bass is a synth and super dark and brooding. Both of those sounds impress me. Mord Fustang's Lick the Rainbow (terrible name I know), but this has quite a few synth sounds that just impress me. Still trying to figure out that bass sound myself. So much breath and life in that. Your synth sounds aren't bad, but if you want something to really show you what is possible with just one synth, then look no further than KVR's One Synth Challenge (I've entered a few myself) and let me tell you some of the sounds people are able to coax out of these synths is simply astounding.