External audio interface - scam, or necessity? (intel mac)

[jonny deep]

I find it useful to do both; I have matched headphones and switch back and forth between monitoring the board and the output signal of the recording DAW, I also use a DR-05 and monitor that via its line out, or at the board, or if that's recording whilst I am also recording the signal to the mac mini simultaneously using Garageband I'll monitor it at the board, the DR-05 or at the output signal of Garageband. That way I can record the same take at 24b 96K in the DR-05 to use later if I really like it. Though I never really like it, which basically led to this contentious question in the first place

I misunderstood a previous post of yours and was trying to make the point that there is still latency introduced by onboard sound. My mistake!

However, it is true that a USB interface with ASIO drivers will give you lower latency than onboard sound. The driver performance is key - it's not just about what's happening in the DAW.

I realise this is all moot, but if you find you want to improve latency (perhaps so you can hear VST instruments in near real time), then an external interface and ASIO drivers are a good way to achieve that.

 

[pazu]

That's cool, I appreciate your advice. I do want to lower latency where I can. If driver performance means better recorded audio, then I need to get up to speed on driver performance. It is true that there is a bigger picture here, it seems clear that external interfaces bring powerful advantages to the overall craft, I am definitely reading that. The ext int's are well worth the investment. I have been looking at the berry mixer built in recorder decks out there now, can put compression on channels, record to SD I believe.

 

[DM60]

As stated in an earlier post, this is not to convince you to buy anything, but intended for further information.

Mac on board sound (PC even as they are usually the same components at a comparable price level, meaning a higher in Windows based machine). It is designed for end user use. It is not a matter of how good it is or isn't, but what it was designed to do. The sound circuitry was not designed with a primary purpose of recording high grade recording. Say what you want say what you will, but that is a fact. I would further add, to maximize profits (they are in business to make money), the system is designed to give the highest quality results for the intended purpose with the lowest cost as possible. With that in mind, your argument could be used for video production, which you wouldn't even think about.

In regards to further arguments, first, the interface does pre-process data (hence AD/DA converters) which reduces the load away from the processor. With an interface, the sound is being processed by the interface, the computer is now just presenting the information to the user and collecting the file information and so forth. On an integrated sound card, by design, it is sharing resources with the motherboard, therefore having more of a load effect to the entire system.

ASIO drivers is an area that changes the equation a lot. When using an on board sound card, you are also using the OS's generic functionality. When they design the drivers for the OS, the design it to meet all of the needs of the end user. A one size fits all. Even if you don't require a part of a functionality, the system is still burdened with that functionality whether you are using it or not. It is a one size fits all. With thr ASIO driver, it bypasses the OS driver and removes a layer of the OS overhead so that the hardware and software can talk directly. Thus reducing processing overhead and the OS middle man. The ASIO drive has one purpose, record and send sound to this one particular device and remove anything it doesn't need. There is more direct communication. That is why there is better latency control from the ASIO driver than the OS driver.

With all of that said, when you record through the input of the sound card, you have to ask yourself, is the file being created the best in level of detail. Take a camera again. A 2MB pixel picture from my phone is not the same as from a nice digital camera. Same file size, but I have seen the difference. There is more to it than just 44.1 sound capture.

At this point, I suggest again, you not buy anything. Record like you are and if the results work out, good for you. But when you want to move up to higher grade sound, just like a camera, you will need to step up.

 

[ecc83]

"Your reference to mac audio being "notoriously poor",

Did not say that. I said OBS electronics was notoriously poor and I have no evidence that macs use better chips than PCs?
I did not mean to be disagreeable, just friendly joshing but will refrain from the B word if it offends!

Take a look at the attachments. My baseline soundcard noise is around -83dBFS* but your clip is at only -63dB ( pk for both) . That frankly is little better than a top line cassette running Dolby B and is about 7dB worse than a 15ips Studer with Dolby A!
You could actually save hard drive space and record at 16bits because the system is barely 11.

I can if you like hook up my NI KA6 which I am confident will better both at some -100dB.

Harmonic distortion is beyond my means these days (I HAD access to an AP test set) but of course I would in any case need a recorded tone. You might be interested to record 1kHz at say -1dBFS and then run it thru a spectrum analyser and see where the sidebands come up to? But in truth, any capacitor mic buffer amp will produce more THD than almost any A-D/D-A converter in this context and your speakers a couple of orders more at 85dBSPL.

*That is at least 7dB worse than it should be. Just shows how things can get away from you if you don't keep up!

Dave.

 

[pazu]

Hold on a second, that made me realize something. There are effects on those two tracks; one is male, the other female. The default. Let me run that again, if you will; I think that we are seeing the effects on the tracks.

 

[pazu]

Sorry about that ecc83. Here is 5 seconds without the effects. Perhaps it is no different, couldn't say myself.

https://app.box.com/s/5w2rogumxfyg6qy5c8rm0ncv7nqh6lo1

I'm not offended by anything, just asking a question, perhaps you've brought it to the definitive answer.

 

[pazu]

I am thinking that the previous two tests were not testing the audio D/A in as much as they were revealing some shortcomings of Garageband 'll as a DAW. Though it works in 24 bit, I believe that the output is 16. Here is a capture run from another piece of software, which should be of higher resolution.

https://app.box.com/s/uakzkkdq80lg9t2avuima0gdfyqr9sum

You will have to forgive me as to not knowing quite what I am sending you; I haven't tested my captures like this before. Thanks for looking, if interested!

 

[ecc83]

That's better!
You have now bettered the A Dolbied Studer (but SR would still tear you a new one!).

Of course, having a -100dB noise floor is somewhat academic, anything you connect to such a system will be vastly noisier but the point is, the design effort needed to get such low figures shows generally good quality. In any case if you were to listen to certain musical forms in a very quiet studio on top line monitors a noise level of -60dB would be audible to some people. |(wish one of them was me!)

It is also a basic tenant of "professional" audio that the studio specifications are much higher than the finished product. Thus we*record at 24 bits, some at 96kHz (tho there is only the flimsiest of reasons to do so!) We demand the highest possible figures for noise, THD and crosstalk from mixers etc....None of which matters a toss for bog S MP3!

Yes, carry on as you are, if your results are acceptable to you and others, no worries. But don't knock USB AIs till you have tried one!

*I use the "royal" we. I do not have any kid of studio, just the remnants of a departed, very musical son. I am registered deaf in both ears but still like the technical! N.B. I still have good enough lugs to enjoy music. Bach is my top choice but I can headbang with the best of them. Love jazz.

Dave.

 

[pazu]

That's better!
You have now bettered the A Dolbied Studer (but SR would still tear you a new one!).

Of course, having a -100dB noise floor is somewhat academic, anything you connect to such a system will be vastly noisier but the point is, the design effort needed to get such low figures shows generally good quality. In any case if you were to listen to certain musical forms in a very quiet studio on top line monitors a noise level of -60dB would be audible to some people. |(wish one of them was me!)

It is also a basic tenant of "professional" audio that the studio specifications are much higher than the finished product. Thus we*record at 24 bits, some at 96kHz (tho there is only the flimsiest of reasons to do so!) We demand the highest possible figures for noise, THD and crosstalk from mixers etc....None of which matters a toss for bog S MP3!

Yes, carry on as you are, if your results are acceptable to you and others, no worries. But don't knock USB AIs till you have tried one!

*I use the "royal" we. I do not have any kid of studio, just the remnants of a departed, very musical son. I am registered deaf in both ears but still like the technical! N.B. I still have good enough lugs to enjoy music. Bach is my top choice but I can headbang with the best of them. Love jazz.

Dave.

I'm grateful for the analysis. Wohh that is a unique position to be in, in terms of music appreciation. I promise I'll stop knocking USB!!!

All a big learning experience for me. I appreciate the detail that posters have gone into. Now I have to learn what these screen shots mean...

Okay this is interesting, I just recorded then analyzed it myself, and I get different results. Actually this time I had the mixing board on.

View attachment 93161

 

[pazu]

Here is the file in the last screenshot which I am also including here:

https://app.box.com/s/p2dydqkm1h1p3w068h2l4n48ikh5qus1

View attachment 93162

I can't say for sure, but thinking that I've a lower noise floor than earlier files indicated.

 

[pazu]

This link is to a short arpeggio on D that I just recorded with an Ibanez AW/Fishman internal pickup via XLR to the berry pmp4000, out the two monitor sends and into the Mac Mini line in, recorded with Adobe Audition CS6. There's plenty of lead in but the guitar is live on the channel.

https://app.box.com/s/b4nsxsasfv25w93c9asadu95d4607g8b

View attachment 93163

 

[ecc83]

Sorry but they are spectrums I showed you the actual noise levels on the DAW meters.

I used the rather ancient Samplitude 8se because not only does it have a 90dB range but also a numerical readout in both rms and peak that goes below 90dB.

A much clearer spectrum can be obtained from the free program "Right Mark Analyser" I have to cook dinner soon but I shall come back in the morning with some fresh data.

Dave.

 

[pazu]

Sorry but they are spectrums I showed you the actual noise levels on the DAW meters.

I used the rather ancient Samplitude 8se because not only does it have a 90dB range but also a numerical readout in both rms and peak that goes below 90dB.

A much clearer spectrum can be obtained from the free program "Right Mark Analyser" I have to cook dinner soon but I shall come back in the morning with some fresh data.

Dave.

Oh! Well I don't know quite 'where' it is being measured apart from in the file but it is my working understanding that this is a graph of dB and frequency, dB being amplitude, and I think that the blue area is basically where there's energy present at a given frequency with no actual wanted signal delivery, I don't know much about it but thought that this would be the 'noise floor of a given 'slice' of recorded bandwidth.

I downloaded Right Mark Analyzer and installed in on XP running on my mac via Parallels and it runs, but something wasn't quite right, it wasn't seeing audio so I haven't got a result there yet.

Thanks for being willing to help, this is about all I can do in terms of audio analysis with what I have for software.

View attachment 93165View attachment 93164

 

[mjbphotos]

1) I'm guessing you're really doing 16bit stuff with the internal soundcard - you may have GB set ot 24bit, but I've never heard of a consumer computer soundcard with 24 bit capability. So running GB in 24 bit mode serves no purpose.
2) If you are getting 'no noticable latency' from your internal soundcard, great - that's not the case with most internal soundcards.
3) I see you mentioning USB hubs multiple times. Common advice for recording - do not use a hub. Plug the rest of your USB devices (keyboard, mouse, etc) into the hub, plug your interface directly into a USB port. If you don't have enough ports, shame on you for buying a Mac! I've got 8 ports on the back of my PC and 2 on the front.

 

[ecc83]

1) I'm guessing you're really doing 16bit stuff with the internal soundcard - you may have GB set ot 24bit, but I've never heard of a consumer computer soundcard with 24 bit capability. So running GB in 24 bit mode serves no purpose.
2) If you are getting 'no noticable latency' from your internal soundcard, great - that's not the case with most internal soundcards.
3) I see you mentioning USB hubs multiple times. Common advice for recording - do not use a hub. Plug the rest of your USB devices (keyboard, mouse, etc) into the hub, plug your interface directly into a USB port. If you don't have enough ports, shame on you for buying a Mac! I've got 8 ports on the back of my PC and 2 on the front.

http://www.hardwaresecrets.com/datasheets/ALC888_1-0.pdf

Sorry Mike, that is what I thought but the above is the manual for the sound chip in my pretty basic HP i3 laptop (smell under £400) and it is quite clear it will record and play 24bits.
I have also done a test recording in Samplitude and I set for 24/44.1kHz and it did it and exported it and I know it is 24 bits because rm anylyser won't open it!

OP: I SHALL get back with more data lata!

Dave.

 

[bouldersoundguy]

2) If you are getting 'no noticable latency' from your internal soundcard, great - that's not the case with most internal soundcards.

Actually, it's the case with every sound card on the planet when used with an analog mixer as a front end. An audio interface is a sound card with a built-in mixer. There seems to be persistent confusion about this among people who have never used a mixer with an audio interface.

 

[mjbphotos]

Actually, it's the case with every sound card on the planet when used with an analog mixer as a front end. An audio interface is a sound card with a built-in mixer. There seems to be persistent confusion about this among people who have never used a mixer with an audio interface.

What?
No latency if you are just direct monitoring the inputs from the mixer - that's no different than direct monitoring from your interface.

 

[bouldersoundguy]

What?
No latency if you are just direct monitoring the inputs from the mixer - that's no different than direct monitoring from your interface.

Right. Using a mixer with a sound card the usual way you don't get input monitoring latency so #2 on your list doesn't apply. That's all I was saying.

 

[pazu]

1) I'm guessing you're really doing 16bit stuff with the internal soundcard - you may have GB set ot 24bit, but I've never heard of a consumer computer soundcard with 24 bit capability. So running GB in 24 bit mode serves no purpose.
2) If you are getting 'no noticable latency' from your internal soundcard, great - that's not the case with most internal soundcards.
3) I see you mentioning USB hubs multiple times. Common advice for recording - do not use a hub. Plug the rest of your USB devices (keyboard, mouse, etc) into the hub, plug your interface directly into a USB port. If you don't have enough ports, shame on you for buying a Mac! I've got 8 ports on the back of my PC and 2 on the front.

Mac on-board sound is 24 bit in the mini & I can do 32 bit floating. GB is always in 24 bit, there's no changing that, only 'sharing' (their term for exporting) at lesser bit rate & resolution. I'm using Adobe Audition CS6, which happens to be mac software. I've got 4 windows boxes actually, one reconfigured as a firewall/router (untangle), one is an 8.1 machine, and two XB boxes. Plenty more in the junk pile - though all of the macs that I've owned, still work. I know PC's since DOS 3.1. I use Macs as well, also have a Raspberry Pi running Linux Mint. And a slew of other weird decks, but platform isn't really the point. What I'm trying to learn, is whether my audio is clearly inferior to an external audio interface.

If my recordings are 16 bit, well then that would be a huge reason - but I do not think that they are. If my headroom is in the 50's, that would be a good reason - but I do not think that it is.

With hope Dave/ecc83 will be able to analyze my later files for noise etc. I Found that I seem to be able to create a lower 'noise floor' (if I'm understanding this correctly and I suspect that I may not yet be), by lowering the input level in the mac. I had hesitated to make an adjustment there, but now think that it might well be worth it.

I think that this is getting sorted out. If I can't achieve the noise floor approaching that of an economical 24 bit external interface, and get other features besides, if that proves the case, then I've a good reason to change. If my machine were only capable of 16 bit recording, well that would have been a good reason to change.

 

[pazu]

Okay, maybe this will put the thread to rest, I will have the definitive answer that I seek, and this noob will go away until my next dumb question - edit - I'm happy to help; it is just that in this topic area I've got a lot to learn. I am not participating solely to tap brains - I would help out or contribute any time, if I had a valid thought to put forward. anyhow..

I am not a 'cover band' sort of guy, but I have recorded a snippet of a 'cover song' here, to provide something solid to analyze - if anyone cares to. Remember, you are defending the modern holy grail of 'external audio interfaces' here; and their manufacturers await, with baited breath...

This is 2-track recording with guitar on the right, vox on the left. Vocal mic is an MXL 990. Guitar is an Ibanez cutaway acoustic with an Ibanez/Fishman SST pickup turned down as far as possible, running wired/XLR balanced into the PMP4000. Preamp gains on the berry are all down. EQ's are flat. I am sending to two montor send outs on the PMP4000, out those to the Mac Mini. You can hear the PMP4000 fans running in the background. Incidentally I don't like the tone of this pickup signal of the Ibanez even with fresh batteries but, whatever, my son is using my only other mic (this is home recording). Also paradoxically, I'm told by open mic people that the 1/4" out of this guitar sounds amazing. Not feeling it myself.

Obviously for purposes of this test recording I haven't applied NR or effects of any kind (far as I know). Levels for L & R were arbitrary (this isn't about seeking the perfect mix). The PMP4000 feeds the Mac Mini's line in, and I recorded this using Adobe Audition CS6, at 32 bit floating, 48K - arguably the highest resolution ever necessary for this sort of thing. I confess to having tried 96K, but I got dropped samples so backed off.

The questions -

What are the specs of this recording, what is the noise floor! ? Can anyone help to determine that?

IS IT EVEN 24 bit? IS it 24 bit depth with the 32 bit floating asides? Is the recorded file actually 48K samples/second? IS IT WHAT I THINK THAT IT IS?

In short, did my system do good work, or will an external audio interface, notably enhance my recordings?

Bonus point for identifying the original artist of the song snippet.

I can't think of anything else to factor in - but I'll answer any questions, and I greatly appreciate any help that you all might provide. I'm just a home recording guy wanting to better understand what's *really* going on relative to my deck vs. adding an ext. audio interface THank you, - Jonny D.

The file: https://app.box.com/s/wocrxx8lkd8sn6k6a8mp7ms07wiogm8l

A screen shot of the file in progress:

View attachment 93189