Hey, that's good enough to be added to Tom's book! I like it... Hey dudes and dudettes, the drummer likes it--it must be good... (giggle) Nick ----- Original Message ----- From: Bryan Smart To: ddots-l@xxxxxxxxxxxxx Sent: Friday, July 24, 2009 4:22 PM Subject: [ddots-l] Re: question about microphone gain input Just wanted to point out that these sorts of situations are one of the reasons why high-end preamps and 24-bit recording is superior to budget pres with typical 16-bit recording. With a budget setup, even with a quality mic, you're always fighting the noise floor (the background hiss/hum/buzz coming from the analog part of your recording setup), and dithering artifacts (the extremely quiet digital noise that is the result of attempting to record signals that are more quiet than the sampling resolution can accurately represent in digital form). Here are a few interesting facts about levels and digital recording. First, about levels. You probably know that the strength of a signal is measured in DB, with 0 being the loudest possible signal that can be accurately stored in digital form. As the number decreases, the level drops off on a logarithmic, not a linear, curve. This is important to know. If, for example, you lower a signal by 6 DB, the signal will sound half as loud as before. For each additional 6 DB, the signal will be half again as quiet as before. The same idea works the other way around: if you raise a volume control in Sonar by 6 DB, the track will sound twice as loud as before. This fact about levels is directly related to how the signal is digitally represented. As you might know, the computer represents all sound data in binary (as a long pattern of 1s and 0s). If you are recording in 16-bit, then the computer uses a group of 16 1s and 0s (or 16 bits) to describe each digital snapshot/sample that it records from your audio interface's inputs. If you are recording in 24-bit, then the computer uses a longer group of 24 1s and 0s to represent each sample that it takes from the audio interface. Regardless of the length of the group of 1s and 0s, the first few bits always describe the portion of the sound that is the loudest, and additional bits describe parts of the sound that are increasingly more quiet. Specifically, the second bit describes the part of the sound that is half as loud as the first bit, the third describes the part of the sound that is half as loud as the second, and so on. Also, each additional bit has a more difficult time at accurately representing its portion of the sound. Even cheap converters will encode the first 12 or so bits accurately, but, beyond this point, the bits are representing sound that is so quiet, that occasionally what should have been a 0 is stored as a 1, and vice versa. These little errors aren't usually a problem, since they're so quiet, that you don't hear the errors in the sound....until you amplify the sound, that is. Even if you have great mics, great preamps, and great converters, there comes a point where there aren't enough bits available in the sample to represent sounds that are quiet beyond a certain point. So, when you reduce the level of the signal coming in by half, that is a gain reduction amount of -6 DB, and that also means that the first bit of the digital sample is no longer used. If you reduce the incoming signal's level by another half, to a quarter of full strength, that is a total gain reduction of -12 DB, and now the first two bits aren't used. If you recorded a signal at this level with only 16-bit resolution, the converters on your audio interface would only be using the quietest 14 of the 16 available bits that you're using to record the sound. It would be very likely that a part of what you recorded at this level was so quiet that it was beyond the lower limit of the converter to represent in only 16 bits. you could digitally amplify/normalize the recording back up to 0 DB, and the signal would sound loud, but it is still only recorded with 14-bit resolution. This is where the digital artifacts come in. When Sonar amplifies the signal, it basically has to promote bit 3 to bit 1, bit 4 to bit 2, bit 5 to bit 3, etc; all of the bits get shifted to the left/up. However, Sonar still only recorded 14 bits of signal for each sample, so it no longer has the last two bits to accurately represent the quietest detail of your recording. You'll hear some small amount of crackle/fuzz because the bits that were less accurately represented have been promoted to represent louder sound than they originally recorded. This can be bad if you have good converters, but, if you have cheap converters, it can be horrible. Remember that I said that many cheap converters don't accurately represent data beyond the first 12 bits or so? Well, if you have one of these, and you normalize the quiet signal as I described above, you aren't turning a 16-bit signal in to a 14-bit signal, you're turning a 12-bit signal in to a 10-bit signal! I hope that this starts to make it clear why 24-bit is better than 16-bit. If you recorded an extremely quiet signal, as we discussed above, and had to boost it by 12 DB, then you'd still be giving up the same two bits. However, if you'd recorded in 24-bit, instead of recording in 16-bit, then you'd be reducing your recording with 24-bit resolution down to 22-bit resolution. This means that a 24-bit signal that is amplified by 12 DB still has dramatically high resolution (about 64 times as much, in fact) than the original unamplified 16-bit signal. So, in summary, recording in 24-bit allows you to dramatically amplify the recorded signal without introducing a lot of grainy digital artifacts. When you record in 16-bit, amplification really isn't a good option. Now, armed with that info, back to recording technique. In budget 16-bit world, you get your levels set for your vocalist by trying to have them sing as loud as possible, and adjusting the gain on the input channels so that the meters never go over about -3 or -4 during that absolute loudest note. As you've discovered, if you attempt to boost the recorded signal after the fact, you increase the level of the dithering noise. It is also tempting to close-mic the performer since, if you don't, you must turn up the input channel's gain in order to get the meters close to -3 or -4, and, the more you turn up the input gain, the noise floor of the preamp also becomes louder. You can get good results when recording at 16-bit resolution, but it takes a lot of work to do it well. If you record too quietly, then you'll pay for it with dithering artifacts when you amplify the recording during the mix. If you record the signal as hot as possible, then you'll avoid dithering artifacts, but you'll run a greater chance of clipping during a really good take. At the same time, if you turn up your preamp to get a good hot level, the noise floor of the preamp comes up and you hear hiss. If you keep the preamp turned down, but close-mic the performer, you'll have mic proximity effects to deal with, and, for vocalists, you'll risk popped Ps, breathing in the mic, and other hazards that can ruin a take. This is even more difficult when you're recording several people/instruments at once, since you have lots of input levels that are near maximum, and could clip at any time. You can use hardware limiters on the inputs to prevent digital clipping, but they'll still color the sound. Basically, you must really work and/or spend a lot of money on gear in order to avoid problems at 16-bit. However, suppose you had a high-end pre, a high-end interface with quality analog/digital converters, and recorded at 24-bit resolution. When you were setting up levels for your vocalists, you could let the levels peek at -10 or lower. Instead of only 3 or 4 DB of headroom, you'd have 10 DB to work with. This means that, in order to clip, someone would need to sing or play a note that was over 2 and a half times louder than the loudest note that they played when you were setting up levels. This is highly unlikely. Once you've finished recording, you'll still need to amplify/normalize the recordings, but, even after boosting the signal by 10 DB, you still will be using a recording with 22-bit resolution. 24-bit recording, while it will allow you to avoid clipping by recording at lower levels and later amplify the recording with almost no noticeable artifacts, won't save you from a noisy preamp. Preamps seem to do an increasingly worse job with cleanly amplifying a signal the more you turn them up. Besides having a nicer tone (which is a concept that it is hard to describe), higher quality preamps can produce higher gain for your converters without yielding a lot of hiss. You're using a Tascam FW1884. The FW1884's built-in preamps aren't that bad, but they put out a lot of hiss when you crank them up. You have some options. First would be to use better preamps. The FW1884 has an ADAT input that will accept another 8 channels of digital signal from another piece of digital equipment. You can get a dedicated high quality preamp/digitizer unit and connect it to the FW1884 through the ADAT port. Presumably, you'd buy a device that has preamps with greater dynamic range. Then, you could crank up the mics without hearing lots of hiss from the preamps. Second, you should reconsider your mics. In the perfect world, you'd use condenser mics to record your vocals. Good condensers will put out a hot signal, and have a very low noise floor, so you won't have to crank up the preamps very far, and, when you do, you'll have to crank them up a long way before you hear any hiss coming from the mic. Unfortunately, condensers are very sensitive, and so they're not as nice when you can't isolate the performers. If you have a vocal booth/closet, then a condenser mic might be something to think about. If you'll be recording several people in the same room, then condensers won't work, since they'll pick up everything in the room, not just what you point them at. Still, even among dynamic mics, there are levels of dynamic range. If you must use dynamic mics, then better preamps are probably the way to go. There is my essay for the week. *smile* Bryan ------------------------------------------------------------------------------ From: ddots-l-bounce@xxxxxxxxxxxxx [mailto:ddots-l-bounce@xxxxxxxxxxxxx] On Behalf Of Stacy Blackwell Sent: Friday, July 24, 2009 2:51 PM To: ddots-l@xxxxxxxxxxxxx Subject: [ddots-l] Re: question about microphone gain input Omar, in addition to the other comments, I have just recorded 4 lead singers in my band with the levels all supposedly set the same. The male vocals did not require much trim during playback, but the female singer, well, I am having to really turn up the trim and now a high hiss is heard when soloed. I think she was scared of popping the mic and held it too far away. Also, if a person sings to the side or across the top (depending on the type of mic) instead of directly into it, the gain and tone can be affected. Trial and error, live and learn. But I do think that singers, male or female, differ in their abilities to "project" their voices which can result in different gains. I have also learned that I should also wear headphones to listen for a constant entering volume from the singer instead of listening to it after the recording. Keep in mind that I don't have a "real" studio and have this equipment mainly for my own personal songwriting and recording, so this was my first time to record other vocalists. See ya, S.B. ------------------------------------------------------------------------------ From: omarbinno@xxxxxxxxx To: ddots-l@xxxxxxxxxxxxx Subject: [ddots-l] question about microphone gain input Date: Thu, 23 Jul 2009 18:00:26 -0400 Hello, I have my vocal mic plugged into my Fireface Soundcard. When one of my female clients records vocals, I end up having to turn up the gain quite a bit. However, I notice when I speak or sing into the mic, I need the gain up only half way. Is it usually the case that gain input on the soundcard or mixer changes from singer to singer, or does it usually stay at the same level. Thanks. Omar Binno Website: www.omarbinno.com AIM: LOD1116 Skype: obinno1 ------------------------------------------------------------------------------ Windows LiveT Hotmail®: Search, add, and share the web's latest sports videos. Check it out. ------------------------------------------------------------------------------ No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.392 / Virus Database: 270.13.27/2258 - Release Date: 07/24/09 05:58:00