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Anyone able to do computer recording, e.g. guitar

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  • Anyone able to do computer recording, e.g. guitar

    While back, I got a focusrite, USB interface box. Thuoght I would try some recording to hear how bad I sound after the fact. The delay absolutely killed the task. Fiddled for hours so that I could record, play back the recorded track while recording another one. REALLY tough to get that delay out.

    My computer is old and about as close to junk as you can get (windows XP, 10 year old dell). Wondering if anyone is using a similar USB box, with a faster computer, getting better results. My brother does lots of mixing using his higher end MAC, but he doesn't record, its all playback so there's no delay problem.

    Read someplace that the Linux drivers, due to their design, are much faster i.e. much smaller delay. Any truth to that, anyone tried recording on a linux box?
    The only good solid state amp is a dead solid state amp. Unless it sounds really good, then its OK.

  • #2
    Check out this forum; https://homerecording.com/bbs/ It's probably the best place to start.

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    • #3
      Originally posted by mikepukmel View Post
      While back, I got a focusrite, USB interface box. Thuoght I would try some recording to hear how bad I sound after the fact. The delay absolutely killed the task. Fiddled for hours so that I could record, play back the recorded track while recording another one. REALLY tough to get that delay out.

      My computer is old and about as close to junk as you can get (windows XP, 10 year old dell). Wondering if anyone is using a similar USB box, with a faster computer, getting better results. My brother does lots of mixing using his higher end MAC, but he doesn't record, its all playback so there's no delay problem.

      Read someplace that the Linux drivers, due to their design, are much faster i.e. much smaller delay. Any truth to that, anyone tried recording on a linux box?
      My DAW is still an XP machine, and have used it with a couple Motu 324 24 Ch I/O interface units, via (4) ADAT 8 Ch fiber optic I/O on the back of a Yamaha 02R mixing desk. I had started using it with Sonar Calkwalk, with the 02R as the active mixer, rather than mixing 'in the box', so to speak. I had turned the living room of my apt into a music room, with Roland V Sessions TD-10 electronic drums, the two guitarists and my self (bass) playing down the wire with various interface boxes from ART (Pro Channel for the Bass), other boxes as preamp to feed the 02R. All of us on headphones, separate stereo mixes via the O2R's monitoring provisions. Later changed to DigiDesign 002 and Protools LE 8.1. Had Waves Diamond for various effects during recording & mixing, and got great results. The 02R has one nice feature on each channel. Variable short delay...0 to around 100mS on each channel, so got nice spacial placement in the stereo mix with that. The DAW was outfitted with Hard Drive caddies, as all recording was done on the extra drives rather than using the System drive. Over time, as the HD's filled up, I kept adding more HD's to continue.

      Of late, after having to pack up my apartment in Gardena and move out to Glendale, with less space here at home, and the band no longer together, I haven't resumed the music endeavors, and do miss that dearly. For a while, I had continued that as a bass player at my church, learning the songs for the coming week's service on a much smaller system to play against. So, there's plenty of ways to get it done. I just picked up an HP ProBook 6570B Laptop with Windows 10, but don't have any music software for it thus far. I'll have to see what's available from Digi as an update from LE-8.1. I forget what the recording thru-put delay was, but as we were all listening live and not on the processing side, there wasn't any delay. Overdubbing wasn't any problem either, that I recall.
      Last edited by nevetslab; 06-14-2020, 04:18 AM.
      Logic is an organized way of going wrong with confidence

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      • #4
        Mike,
        I'm not sure where you are experiencing the delay (called 'latency' by the digital audio nerds), but I can tell you that all systems will show this delay if you are playing into the converter, processing the sound in the computer, sending that signal back through the converter to your monitoring system for you to listen to. I've learned to NOT try and record that way as it really does mess with the brain's ability to coordinate the music making effort.
        Overdubbing shouldn't be a problem with any system. If the overdubs are out of sync with the previous track, they can be slid by as many milliseconds as needed to align them. Some recording software will allow you to save that number in its "settings" so you only do it once.
        In my experience (not guaranteed to be similar to yours) adding more RAM to the computer helped to speed things up tremendously. My computer was a pretty nice unit back in 2006 when it was new, and it's still somewhat sufficient to the task. I do a lot of recording on it.
        If it still won't get loud enough, it's probably broken. - Steve Conner
        If the thing works, stop fixing it. - Enzo
        We need more chaos in music, in art... I'm here to make it. - Justin Thomas
        MANY things in human experience can be easily differentiated, yet *impossible* to express as a measurement. - Juan Fahey

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        • #5
          Thanks Everyone. Sounds like I just am not using the tools correctly, or as good as they can be. A friend who does this kind of stuff as well, has an old yamaha (think its a yamaha) digital recording system. Says its pretty good despite being so old.

          John, thanks, googling . . .

          Nevestlab, thanks, ahh that is what another friend has, old system like 10+but works well. The MOTU thing is a plugin board right into the computer I think I don't understand all the "stuff" but it has a device driver, but the sound has to make it to the OS where their software is running. I think that is where the delay is, but its smaller with a plugin board than a USB box?

          Eschertron, yeah I tried stupid things at first: use the monitor off the software rather than right off the USB board. That was pure torture. So, I flipped the whatevr swtich so I had the monitor (to my guitar amp) not go through software. So, then I could hear what I was playing without a delay. So far so good. But next thing was to play while listening to a previously recorded track, recording the second track. With a whole bunch of fiddling I got the delays fixed up so that two tracks are ... meh. Sometimes a little glitch here and there. But more than two, and it seems to get worse and worse. And my brain isn't what it used to be.

          Adding ram, my old dell is near the max. You'd laugh, but I think it has 2.5 gig now. May be able to upgrade to 4.

          Started looking at motherboards I could graft into the old Dell but nothing else in there could be salvaged. Power supply, memory, video card, all have to be upgraded, so all Id have left is a case that the new parts won't fit into anyway, since the old MB is some proprietary thing.

          maybe some old plug in cards cuold be had on ebay?

          Thanks for all the suggestions, googling . . .

          The only good solid state amp is a dead solid state amp. Unless it sounds really good, then its OK.

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          • #6
            I would think that there would be software that communicates with the Focusrite interface that allows buffer adjustments to cure any latency. That is the way I have done it with all the computers I have recorded with but perhaps the Focusrite is different. From reading on their website there is a thing called Direct Monitoring. Apparently it states:
            "Typically, your recording software also has the ability to monitor the signal as it comes through. If direct monitoring is being used, software monitoring within your DAW should be disabled to avoid hearing the signal twice – once from the direct monitoring and again from the output from the recording software. This is because you hear the direct sound (before it is passed to the computer) and then you hear the same audio that has been processed by the computer afterwards."

            So to me it does not sound like a true latency problem with the computer but more like two outputs at the same time.
            Applies to: Scarlett & Saffire range With Direct Monitor engaged (Scarlett Solo, Scarlett 2i2, Scarlett 2i4), “Zero Latency...
            When the going gets weird... The weird turn pro!

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            • #7
              Originally posted by mikepukmel View Post
              Thanks Everyone. Sounds like I just am not using the tools correctly, or as good as they can be. A friend who does this kind of stuff as well, has an old yamaha (think its a yamaha) digital recording system. Says its pretty good despite being so old.

              John, thanks, googling . . .

              Nevestlab, thanks, ahh that is what another friend has, old system like 10+but works well. The MOTU thing is a plugin board right into the computer I think I don't understand all the "stuff" but it has a device driver, but the sound has to make it to the OS where their software is running. I think that is where the delay is, but its smaller with a plugin board than a USB box?

              Eschertron, yeah I tried stupid things at first: use the monitor off the software rather than right off the USB board. That was pure torture. So, I flipped the whatevr swtich so I had the monitor (to my guitar amp) not go through software. So, then I could hear what I was playing without a delay. So far so good. But next thing was to play while listening to a previously recorded track, recording the second track. With a whole bunch of fiddling I got the delays fixed up so that two tracks are ... meh. Sometimes a little glitch here and there. But more than two, and it seems to get worse and worse. And my brain isn't what it used to be.

              Adding ram, my old dell is near the max. You'd laugh, but I think it has 2.5 gig now. May be able to upgrade to 4.

              Started looking at motherboards I could graft into the old Dell but nothing else in there could be salvaged. Power supply, memory, video card, all have to be upgraded, so all Id have left is a case that the new parts won't fit into anyway, since the old MB is some proprietary thing.

              maybe some old plug in cards cuold be had on ebay?

              Thanks for all the suggestions, googling . . .
              The Motu system used a computer interface card with firewire to run the two 24 channel rack mount I/O units. Drivers were either downloads or from a disc...can't remember now, as it's been years since I had put the system together.
              Logic is an organized way of going wrong with confidence

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              • #8
                I'm not an expert on this but from what I can see most of the latency (delay) comes from the analogue to digital converter in the audio interface (Focusrite in this case). Units with a better engineered ADC exhibit less latency. (You usually get what you pay for)

                Looking at the computer side of things there a few things that you can do to minimise lag and dropouts like a faster processor, adequate RAM, turning off unwanted services in Windows, and removing unecessary background processes and programs that load when Windows starts. All these things make for a much better result.

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                • #9
                  I suggest having a look at building a dedicated Digital Audio Workstation based on Linux. There are popular distributions like AV Linux that create a working DAW configured ready to go straight out of the box. A simple matter of downloading the ISO and burning an installation DVD or better still on a bootable USB drive. USB install is much faster and the method that I normally use.

                  More info here - https://www.makeuseof.com/tag/6-linu...cians-editors/

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                  • #10
                    There are tradeoffs with any system. Setting the recording buffer to a smaller number, say 512 samples or less*, reduces the latency to 'good enough' to overdub to. But the chance of glitches (dropouts) or the number of glitches once the threshold of CPU usage has been passed goes up. What was said about optimizing the OS for the DAW functions is absolutely true, and indispensable to making the DAW behave well. I'm sure I'm not alone in having spent days - and hours in front of google - reading about the best practices for my particular hardware.

                    Other optimizations I've made including having the disk I read/write for the audio NOT be the system disk, so that system disk access won't halt the recording writes. Also the read/write audio is on a SSD to make that access as fast as possible.

                    If you have an XP machine, looking into Linux might be worthwhile. I've no first-hand experience, but in theory the Linux OS will use way less of the CPUs resources than Windows, freeing up that much more power for your DAW.

                    So depending what's in your budget, getting a decent SSD as a "scratch disk" for audio might solve a few problems. That was my first response when I put the last system together. And Linux can be a free or nearly- free solution to getting more CPU power. Depending on the DAW software needed, of course.

                    *512 samples at 44.1kHz results in just under 12 milliseconds latency.
                    If it still won't get loud enough, it's probably broken. - Steve Conner
                    If the thing works, stop fixing it. - Enzo
                    We need more chaos in music, in art... I'm here to make it. - Justin Thomas
                    MANY things in human experience can be easily differentiated, yet *impossible* to express as a measurement. - Juan Fahey

                    Comment


                    • #11
                      Originally posted by Wal_zz View Post
                      I'm not an expert on this but from what I can see most of the latency (delay) comes from the analogue to digital converter in the audio interface (Focusrite in this case). Units with a better engineered ADC exhibit less latency. (You usually get what you pay for)

                      Looking at the computer side of things there a few things that you can do to minimise lag and dropouts like a faster processor, adequate RAM, turning off unwanted services in Windows, and removing unecessary background processes and programs that load when Windows starts. All these things make for a much better result.
                      Using a separate HD (not using the System Drive) for recording on helps a lot. If externally fed, going by way of firewire would be suitable. USB is too slow. Internally mounted in the mainframe is best.
                      Logic is an organized way of going wrong with confidence

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                      • #12
                        Originally posted by nevetslab View Post

                        ...going by way of firewire would be suitable. USB is too slow. Internally mounted in the mainframe is best.
                        Thunderbolt 3, baby.

                        If I have a 50% chance of guessing the right answer, I guess wrong 80% of the time.

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                        • #13
                          I have been using the old EMU sound cards since 2003. These sound pretty good and work well on older computers. I have used them on systems with XP and newer computers running windows 7. The software is very easy to configure and they are powered by steinberg. That is to say steinberg was the predecessor to creative sound blaster which took them over. Same company but lets just say the newer sound blaster stuff is not as well made. These are great consoles to record at home.

                          https://www.ebay.com/itm/Emu-1820-Di...sAAOSwSWde46HB
                          When the going gets weird... The weird turn pro!

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                          • #14
                            Originally posted by DrGonz78 View Post
                            I have been using the old EMU sound cards since 2003. These sound pretty good and work well on older computers. I have used them on systems with XP and newer computers running windows 7. The software is very easy to configure and they are powered by steinberg. That is to say steinberg was the predecessor to creative sound blaster which took them over. Same company but lets just say the newer sound blaster stuff is not as well made. These are great consoles to record at home.

                            https://www.ebay.com/itm/Emu-1820-Di...sAAOSwSWde46HB
                            That's a handsome interface, that's, um... if I'm allowed to say that
                            If it still won't get loud enough, it's probably broken. - Steve Conner
                            If the thing works, stop fixing it. - Enzo
                            We need more chaos in music, in art... I'm here to make it. - Justin Thomas
                            MANY things in human experience can be easily differentiated, yet *impossible* to express as a measurement. - Juan Fahey

                            Comment


                            • #15
                              While a novice at electronics I earned my bread being a recording engineer for many years so potentially something I can help with!

                              Yes, if you are monitoring THROUGH your DAW you will always arrive at some latency, down to 64 or 32 samples for me on Logic but that requires a separate record HD, a lean operating system and a fairly small project. At least, just recording audio and not running any plugins until after you've finished recording. This 32 samples, best case, latency is fine for playing at home and good enough for most. In the studio it's NOT considered acceptable. A very good drummer or bass player would feel that few milliseconds of latency and it would mess them up. They may not be able to know why but it stops them from giving the sort of unbelievably locked-in feeling. Steely Dan style records are out of the picture!

                              For all critical situations we monitor separate to the DAW. This can be from an analog mixing desk, or a good digital one, a tape machine with a sync head or a feature of the interface. The musicians hear themselves back as fast as the electricity can travel and this is considered best case scenario. This was a deal breaker back in the late 90's early 00's, as no computer was anywhere near fast enough to achieve professionally useable software latency (arguably, this is still the case now), and it meant you needed a desk to route monitoring back to the musicians. Pro tools was the first DAW to 'fix' this, and it's why they stayed with their super expensive proprietary gear for so long (it's still expensive now, but the pro tools approach has more or less met in the middle with other companies interfaces approach these days). In those pro tools interfaces when you routed an input to a musicians monitor output so they could hear themselves it actually physically changed connections inside the interface so they were hearing it more or less analog, you could record in the protools HD DAW and do what you wanted as the engineer, the signal was being controlled and perhaps adjusted for level etc. and then physically routed back to the performer. They went further, adding super fast cards that held effects and so on, so the engineer could add a compressor and the musician would hear it back with only 1 ms or less of latency, even though it looked like a normal plugin to the engineer. (modern DAW compressors have achieved this 1ms or so latency without proprietary gear)

                              If 1ms sounds crazy small, bear in mind that when editing for a commercial release we regularly use steps of 3ms to fit a bass guitar note into the pocket of a kick drum's audio, usually the bass needs to be a little behind the kick to feel 'in time', but usually only by a few ms. Great bass players find that spot automatically!

                              So early protools HD really was a pretty clever system that for a few years was the only way of seriously recording on a computer, hence insane price, my studios system was about 45k

                              But that principle of course worked it's way down, many more interfaces started adding ways of monitoring direct and sending a feed to the software. This is a little like how a tape machine works on the sync head, the musicians audio is sent back to them in real time while the recorded audio that takes time (due to the physical speed of the tape) gets recorded. Later when you don't need to worry about the musicians you switch to the repro (reproduction) head which is a little higher fidelity and less worn out, but its delay is so large it's unusable to monitor off of. (but of course, gave us the tape delay effect)

                              So more expensive interfaces allow similar to the early protools style routing now, but also many budget interfaces do a great job too. I'm not sure what focusrite you have but many of the focusrite interfaces have 'monitor' control. This is a linear control that sweeps between hearing 'just the DAW' to 'just the inputs'. its a very slick way of achieving what you need. Of course, if you go extreme one way or the other you don't hear the track you are recording to/don't hear yourself. So usually around the middle is good. You also need to make sure you DON"T monitor from the software at the same time, otherwise you'd hear both, the DAW a little behind the direct which usually causes a phasey chorusy sound, depending on how behind your daw is.

                              So, check your sound is being recorded ok in the DAW, switch to monitoring directly and turn down the DAW monitor send for recording your actual take. Many DAW's have options to accommodate for this automatically. EG. they allow you to monitor from the software, or turn it off, or some kind of dynamic in between that adjusts automatically when you are actually recording or similar.

                              All through this process, the computer KNOWS how much latency it is applying, it KNOWS the track its playing you is 4ms behind, and it knows what it's hearing from you was played 4ms ahead from when it received it. So when you press stop it shifts things to the right place. On older systems this system could fail, the interface not correctly reporting the delay to the DAW etc. so this is when you need to measure the resulting latency and adjust a manual slider (in Logic, for example, telling it the manual amount to move recorded audio after pressing stop). These days though, they are usually good and sync up fine without user intervention. If you feel your system might be messing up and putting things in the wrong place let me know and I can run through some fixes!

                              Lastly, we get to the more tricky area of plugin latency. The best advice is record with as few plugins as possible, ESPECIALLY avoid limiters with lookahead, clever dynamic noise reduction things and so on. Compressors and eq's are usually JUST acceptable. If you want to know why I'll summarise here..

                              All plugins cause latency. DAW's add up this latency (as long as it's being reported correctly) and align all tracks to be in time with the channel-strip/audio-track thats reporting the MOST latency. This is plug in latency compensation. It works great without any input when mixing. If you are recording then you have a choice -

                              Turn OFF plug in latency compensation; now all your tracks with plugins go out of time a little bit and the groove is muffled maybe. But as long as your monitoring situation is good you can record and no weirdness will happen. But then you turn latency compensation can ON, and it shuffles your track later in time to align with the slowest track that you recorded too. Not ideal and a bit messy!

                              Record WITH plug in latency compensation; Now everything seems fine and you can record to a tight groove thats all nicely aligned. When you press stop the DAW see's that your fresh newly recorded track with no plugins is presenting NO latency, so it shifts it later in time in line with the slowest channel strip. But, you RECORDED to that channel strip? So now your audio is being shifted even LATER than the thing you were recording too??

                              There is no elegant solution, until we get DAW's that dynamically read the plug latency compensation while a track is being recorded. This is fairly conceivable as a simple idea, but bear in mind that many projects might have plugins switching off and on throughout a mix so the latency situation might change throughout, causing the need for the recorded audio to be satisfactorily warped here and there throughout the record-pass . DAW's are getting closer to being able to do this, but as of now the rule is 'save adding any high CPU or fancy plugins until after recording is done'. And watch the DAW forums and manuals for updates, as we are nearing a time when this problem may be fixed... Note I am simplifying massively, how DAW's deal with plug in delay compensation is a minefield and different with each different software version / each DAW.

                              Anyway - I'm throwing out a tonne of info because this may help some peeps and this forum has been awesome for me asking random questions and receiving great answers! The bottom line is try and listen back to yourself DIRECT during recording, especially if your computer is not lean and mean. If it IS, and this includes external drives and clean operating system, then you can probably get away with it on the lowest latency setting (sometimes called buffer size, its the processing buffer). If not, use a little mixer or some way of sending yourself a feed pre-interface, just for listening back to the audio you are recording. Check your interfaces manual for any kind of hardware monitoring feature they have built in to remedy this, as even entry level interfaces these days have sometimes addressed the problem. If your computer behaving weird and moving things around after recording this can be bad reporting of system latency to the DAW, odd DAW settings or plugin delay latency, I guess repost if it is and I can try help!

                              I don't think you need a linux machine, many many have useable rigs on crappy old dells, hell, my first rig was the same. You CAN go the linux route, and it could get you working if you are passionate about building a linux system etc! But all the problems you have came across are age old, normal issues of any computer audio setup, whatever the format, and things I have been employed to fix in any combo you can imagine over the years
                              Last edited by OwenM; 06-16-2020, 01:43 AM.

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