Phil O'Keefe Posted September 17, 2012 Author Share Posted September 17, 2012 My soundcard is an M-Audio Delta 1010. I'm using an old PC from 1999 that runs on Win98SE and Cubase VST. The setup has been dedicated all these years without any updates, so it's really stable. I don't use any midi with it, I just track audio for my demos. It doesn't even have usb 2.0, so I have to save to CD-R to transfer files If you have an open PCI slot, you can always add a USB 2.0 or Firewire card to it. For the sake of backups, I'd highly recommend that. I used to have hair... backing up to CD-R and DVD-R is what gave me my current hairdo. As far as the M-Audio Delta 1010, is it this one? If so, it is still currently available, which means there should be drivers for Win XP / Vista / Win 7. It also supports 24 bit recording at sample rates up to 96 kHz, so unless you have an older version of it that doesn't support 24 bit, there's no reason I can see why you shouldn't record at 24 / 44.1. Windows 98se's FAT 32 file system has a max file size limit of 4GB. You might run into that if you're recording long live concerts, but for the typical pop / rock / jazz / country song, it's a complete non-issue, even at 24 bit. There's also the partition size limits of Fat 32. In a nutshell, you can use a larger HDD, but you'll need to partition it with FDISK into partitions of about 125GB or smaller. The system's stable, but in order to extend its usefulness and lifecycle, I'd recommend: 1. Installing a PCI USB 2.0 card 2. Doing a complete system-wide backup. 3. installing a larger audio-only drive and partitioning it. 4. Changing your settings on the Delta and DAWs to 24 bit. 24 bit recording will gobble up HDD storage space faster than 16 bit. It uses 1/3 more storage, assuming the sample rates and track counts remain the same. It will also put a bit more stress and strain on your older CPU - not really in recording it or playing it back (that's largely a function of the drive seek and throughput speeds), but in terms of CPU processing for things like plugins. If you're not using a ton of plugins, you should be fine... but if you've been pushing your old CPU to the limit of its capabilities with them, you will probably run into issues at 24 bit with the same amount of plugins. The basic audio quality will take a significant jump upwards at 24 bit though, and IMHO, it's more than worth it. Link to comment Share on other sites More sharing options...
Members Fender&EHX4ever Posted September 18, 2012 Members Share Posted September 18, 2012 If you have an open PCI slot, you can always add a USB 2.0 or Firewire card to it. For the sake of backups, I'd highly recommend that. I used to have hair... backing up to CD-R and DVD-R is what gave me my current hairdo. As far as the M-Audio Delta 1010, is it this one? If so, it is still currently available, which means there should be drivers for Win XP / Vista / Win 7. It also supports 24 bit recording at sample rates up to 96 kHz, so unless you have an older version of it that doesn't support 24 bit, there's no reason I can see why you shouldn't record at 24 / 44.1. Windows 98se's FAT 32 file system has a max file size limit of 4GB. You might run into that if you're recording long live concerts, but for the typical pop / rock / jazz / country song, it's a complete non-issue, even at 24 bit. There's also the partition size limits of Fat 32. In a nutshell, you can use a larger HDD, but you'll need to partition it with FDISK into partitions of about 125GB or smaller. The system's stable, but in order to extend its usefulness and lifecycle, I'd recommend: 1. Installing a PCI USB 2.0 card 2. Doing a complete system-wide backup. 3. installing a larger audio-only drive and partitioning it. 4. Changing your settings on the Delta and DAWs to 24 bit. 24 bit recording will gobble up HDD storage space faster than 16 bit. It uses 1/3 more storage, assuming the sample rates and track counts remain the same. It will also put a bit more stress and strain on your older CPU - not really in recording it or playing it back (that's largely a function of the drive seek and throughput speeds), but in terms of CPU processing for things like plugins. If you're not using a ton of plugins, you should be fine... but if you've been pushing your old CPU to the limit of its capabilities with them, you will probably run into issues at 24 bit with the same amount of plugins. The basic audio quality will take a significant jump upwards at 24 bit though, and IMHO, it's more than worth it. Great info. No pun intended, but it's a lot to process right now I will definitely look into improving the setup with this info Link to comment Share on other sites More sharing options...
Members flatfinger Posted September 19, 2012 Members Share Posted September 19, 2012 Try it mikey ; you'll like it !!!! http://klanghelm.com/VUMT.html Link to comment Share on other sites More sharing options...
Phil O'Keefe Posted September 20, 2012 Author Share Posted September 20, 2012 Try it mikey ; you'll like it !!!!http://klanghelm.com/VUMT.html That looks interesting! I noticed you appear to have single VU meter pictured in your avatar - did you have a hand in developing the plugin? Link to comment Share on other sites More sharing options...
Members gubu Posted September 20, 2012 Members Share Posted September 20, 2012 Ok, I've got another question about this:- I'm still not 100% sure what is meant by 'average' signal in the doc in the OP, and in this thread. For instance, I'm playing back a stereo drum recording that is peaking at -1.9dBFS. But the RMS (average) meter is hovering around -20dBFS and below. So, if I trim this channel, the RMS will get even lower. Are we talking average peaks that you measure with your eyes on a digital peak meter, or RMS measurements, when we say 'average' levels at around -15dBFS? Link to comment Share on other sites More sharing options...
Phil O'Keefe Posted September 20, 2012 Author Share Posted September 20, 2012 IMO, your peaks are dangerously high, and so in that case, I would recommend trimming it a bit lower. I am recommending keeping the average levels (RMS) at around -15dB, or a bit lower. If you're working with highly dynamic material, then you have a choice - lower the average levels to avoid peaks and clipping, or use a compressor / limiter to reduce the dynamic range. Typically, I would tend to opt for the former as a matter of default as opposed to the later. The reason is, while I may sometimes want to compress something for artistic / sonic reasons, I don't want to have to compress - sometimes I want it recorded as cleanly as possible. Link to comment Share on other sites More sharing options...
Members gubu Posted September 20, 2012 Members Share Posted September 20, 2012 Ok, thanks Phil. That's much clearer. And is what I have been doing all along, i.e. no peaks above -6dBFS. Link to comment Share on other sites More sharing options...
Members flatfinger Posted September 21, 2012 Members Share Posted September 21, 2012 That looks interesting! I noticed you appear to have single VU meter pictured in your avatar - did you have a hand in developing the plugin? No , I just read Bob Katz book about mastering a few years back and got away from watching digital peak meters .....( Oh Golly don't I miss that !!!!!) I discovered the Klanghelm plugs over at KVR P.S. This dev also has a really cool , reasonably priced compressor to! Link to comment Share on other sites More sharing options...
Members flatfinger Posted September 21, 2012 Members Share Posted September 21, 2012 I also paid allot of attention to Paul Frindell... Here's a snippet from him at another forum ... here are 2 reasons to record, process and master at less than flat out dBFS: 1. To avoid hidden overs not shown on metering due to the reconstruction of the signal. This occurs mostly in D/As (variably depending on how they are designed) and to some extent within some plug-ins and digital processes. Error from this range from limiting all the way to loud 'splats' as values may fold over completely. A safe margin for normal programme to avoid most of this is -3dBFS. Although some artificially maximised programme can create more than this and some test material can create 6dB of over - so real safety is only gained at around -6dBFS. If you have a reconstruction meter you can monitor this effect yourself (for the mix output) and compensate manually, or if you have a suitably equipped limiting app (I.e. Oxford Limiter) you can correct these errors automatically. This does not help much with stuff within the channels of the mix itself - so prudence us still advisable. 2. To create headroom, using lower levels within your mix allows you to avoid clipping signals every time you do anything - it frees you up to concentrate on sound rather than red lights and radically eases the mixing process. Some plugs may actually sound better because internal overs may be avoided. To do this you need to reduce levels to something sensible first thing in the playback channel - process at lower levels - end up with a mix at less than flat out - then make up the level at the very end of the mix. The part in bold italic underline ( enough emphasis ??) is why I love the :love:VU !! Link to comment Share on other sites More sharing options...
Members flatfinger Posted September 21, 2012 Members Share Posted September 21, 2012 http://www.soundonsound.com/sos/dec11/articles/qanda-1211-5.htm . Link to comment Share on other sites More sharing options...
Members Y0UNGBL00D Posted September 26, 2012 Members Share Posted September 26, 2012 im always worried my mixes are way too quiet, but i defeinitely work on that side of the line. Link to comment Share on other sites More sharing options...
Members Zooey Posted September 26, 2012 Members Share Posted September 26, 2012 im always worried my mixes are way too quiet, but i defeinitely work on that side of the line. Tracking hot will not make your mixes louder anyway. Link to comment Share on other sites More sharing options...
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