|
3 Questions, regarding bit depth & mono/stereo stuff,
|
View this Thread in Original format
| Damnuok2 |
Hey guys thanks for wasting your time answering to my questions...
1st:
I've always thought it doesn't matter if you work with 16 bit or 24 bit samples, cause when you finish your track/burn it to CD etc, it will end with 16 bit, but today I heard one famous producer saying that it was better to work with 24 bit samples cause when you're rendering your bassline, leads, etc from the synths they will be rendered to your's DAW default bit depth (24/32 whatever), so it will mess with ur 16 bit drum/fx samples?? This is a bit confusing...can anyone try to explain to me if it makes any sense?
(btw I've heard this at this video: [[ LINK REMOVED ]]
and it's a really nice video)
2nd:
I've heard many people saying that all the 'low' elements (kick,bass...) should be mono, but I've also heard many people saying otherwise...which one is the most correct view?
I know there are some basses that sound better in stereo and some in mono, but stereo basses won't get ed up in a club mono system, even if they sound right on your studio?
Also, a question to the ableton users, to make tracks (like the kick) mono, we must use the utility plugin, stereo with 0% width?
3rd:
I've also heard guys saying that its better to pan your drums a little/add a bit of stereo image, even if the chance isn't noticeable at all, because in the end, when you're at the famous mastering process, it will sound better? ([[ LINK REMOVED ]]
)
Sorry for the silly questions, but I really have to clean up my mind :)
Thanks, again. |
|
|
| palm |
1:
work with 24bit, u get better dynamics. Render master at 24bit maybe 48kHz wav then let the label master it and encode it to mp3 16bit 44,1kHz.
2:
I usualy mono everything below 60Hz caus it sounds stupid on a club if the bass is diferent on the speakers, it will up the balance. also if your label is releasing vinyl its even more important beacuse the vinyl might start skipping, needle jump, if bass is in stereo, also here this is labels work.
3: pan your drums if u want to but keep in mind that if u pan one drum to the left you should have a counter drum on the right so it balances good. also try mono your track once in a while to make sure it sounds good then too (elements must not outface each other imo).
In the end you shouldnt worry too much about stuff like this as most of this is labels job, just try to keep the track as clean and pro as possible. IMO |
|
|
| Zombie0729 |
| 2. use the utility plug and set the direction to : Left... there is your mono signal |
|
|
| derail |
The 1st question I also need to get my head around at some point. Maybe it's explained in the Bob Katz book...but I hear often that a higher bitrate results in better quality when going down to 16 bit, 44.1 kHz CD quality audio, whereas higher samplerates don't matter at all when going down to 16/44.1.
The way I think about it is, with a samplerate of 44.1 kHz, it's essentially taking 44,100 "snapshots" of the waveform per second, and those snapshots are going to be the same at any given point in time, no matter whether the original is at 44.1 kHz or 48,96,192 or 384. I think it only matters if the track stays at the higher samplerate.
However, with bit rates, that relates to the number of degrees of loudness you can have in the track. So if you have decent equipment, you may hear things like reverb tails fade more smoothly if you're running at 24 or 32 bit as opposed to 16 bit. The reason I think bitrate matters more (and I could be way off on this...) is that when you take a whole bunch of 16 bit sounds and add them together, you could well end up with a difference in volume, compared with adding 24 bit sounds together...when it gets converted to 16 bit, it may well round to a different number/overall loudness level.
If there's anyone on this forum who can explain it in simple layman's terms, that would be great! My reasoning may be totally off. I really have to do some reading...but a lot of it is reasonably technical...
The 2nd question - historically, because of the vinyl thing, bass frequencies were all converted to mono at the mastering stage (or beforehand, since people knew it'd be converted later anyway). So a lot of people grew up hearing a mono bass - when you hear every song doing that, it just "feels right". It really stands out when the low end is very stereo, since not too many songs have that. I'd say another really good reason to keep a bass mono, for trance, is as stated above, that if people are listening to the song in a club, and the bass has a stereo effect on it, then it's going to get louder and quieter in each speaker (if people are standing directly in front of one speaker, they're not going to get a stereo image. They'll just hear that sometimes the bass seems to fade away, then come back...rather than being consistent). But apart from those reasons, feel free to widen the bass. If it's for headphone listening and it sounds good, then it's fine. I've tried it a few times and it never really hits my ears in a pleasing way. Usually if people want to create a wider bass, they don't apply the widening to the really low frequencies. So it still sounds wide, but not "funny/weird wide". (Yes, the utility plug in Ableton will make sounds mono. Then, to apply widening, use an fx bus with maybe a chorus, or delay, or whatever, followed by an eq/ high pass filter, so you can remove the low frequencies from the effect)
3rd question - This is much more up to your personal preference. Sometimes great tracks have a hihat sticking out like crazy all the way over to one side and it sounds fine. Sometimes everything's very centred and it sounds fine. I guess in general terms, songs have them spread out a bit (but not panned hard left or right - generally things like pads are the widest elements - generally...), and generally they're also balanced, so you have about equal amounts happening in both ears. But there are many instances of fantastic tracks where this isn't the case. Listen to your favourite productions and see if you have any personal preferences, or just approach it on a song by song basis. |
|
|
| Zombie0729 |
| quote: | Originally posted by derail
The 1st question I also need to get my head around at some point. Maybe it's explained in the Bob Katz book...but I hear often that a higher bitrate results in better quality when going down to 16 bit, 44.1 kHz CD quality audio, whereas higher samplerates don't matter at all when going down to 16/44.1.
The way I think about it is, with a samplerate of 44.1 kHz, it's essentially taking 44,100 "snapshots" of the waveform per second, and those snapshots are going to be the same at any given point in time, no matter whether the original is at 44.1 kHz or 48,96,192 or 384. I think it only matters if the track stays at the higher samplerate.
However, with bit rates, that relates to the number of degrees of loudness you can have in the track. So if you have decent equipment, you may hear things like reverb tails fade more smoothly if you're running at 24 or 32 bit as opposed to 16 bit. The reason I think bitrate matters more (and I could be way off on this...) is that when you take a whole bunch of 16 bit sounds and add them together, you could well end up with a difference in volume, compared with adding 24 bit sounds together...when it gets converted to 16 bit, it may well round to a different number/overall loudness level.
If there's anyone on this forum who can explain it in simple layman's terms, that would be great! My reasoning may be totally off. I really have to do some reading...but a lot of it is reasonably technical...
|
taken from Rick Snowman:
"All computers utilize the binary language that consists of two values, 1 and 0. The computer can count up to a specifc number depedning on how many bits are used. For example, using an 8-bit system it is possible to count to a maximum of 255, while in 24bit system the maximum value is 16777215.
Relating this to digital audio recording system, the number of bits determines the number of analogue voltages that are used to measure the volume of a wave form, in effect increasing the overall dynamic range. In technical applications the dynamic range is the ratio between the residual noise(known as noise floor) created by all audio equipment and the maximum allowable volume before a specific amount of distortion is introduced.
...
...
...Essentially, this means that if a low bit rate is used for a recording, only a small dynamic range will be achieved. The inevitable result is that the ratio between the noise floor & the loudest part of the audio will be small, so background noise will be more evident."
hope that was some help guys, thats how i have always understood it |
|
|
| Eldritch |
| quote: | Originally posted by derail
However, with bit rates, that relates to the number of degrees of loudness you can have in the track. So if you have decent equipment, you may hear things like reverb tails fade more smoothly if you're running at 24 or 32 bit as opposed to 16 bit. The reason I think bitrate matters more (and I could be way off on this...) is that when you take a whole bunch of 16 bit sounds and add them together, you could well end up with a difference in volume, compared with adding 24 bit sounds together...when it gets converted to 16 bit, it may well round to a different number/overall loudness level.
|
That is incorrect. 24 bit isn't louder compared to 16-bit, it's just more detailed.
However, you can record louder relative to the noise floor with 24-bit. The benefit of using 24-bit isn't always that huge. But it's a good idea to use 24-bit anyway.
By the way, bitrate is the wrong term. Bitrate refers to bits per second, i.e. file size.
Bit depth is the correct term. |
|
|
| derail |
| quote: | Originally posted by Eldritch
That is incorrect. 24 bit isn't louder compared to 16-bit, it's just more detailed. |
I didn't say that 24 bit was louder, but that "the sound may round to a different loudness level". It may round down and be quieter, it may round up and be louder. When I referred to "the number of degrees of loudness", I meant exactly what you meant by "more detailed". It's just a case of linguistics.
Yes, bit depth is the correct term. I stand corrected. |
|
|
| G-Con |
Just a quick question over something in this thread that has me confused.
The op asked if he should put the utility on tracks he wants to be mono.
I always have my kick and low bass mono, but I don't put the utility on them. I just make sure not to apply any widening effects of any kind - chorus, stereo delay, unison spread etc.
Is this not proper mono? Should I be applying the utility plugin to my bass and kick? |
|
|
| palm |
if the samples itself is in mono it should be ok. but your synths, reverbs etc etc might have information below 100Hz in stereo so I use the stereo expander (this is in reason) with a cutover freq at around 80hZ and mono everything below, now this cutover works as a 6dB/oct filter so its not suddenly mono below 80hZ, its cradualy more and more mono the lower u go. i dont know if there similar tools in ableton etc.
btw i try not to use one device on master as it seems to affect the sound abit, i use one separate for each track (most of the time). |
|
|
| derail |
| G-Con, it's fine to leave it as a stereo sound if there's no stereo spread to the sound. A lot of stereo bassdrums and basses are actually mono, just in a stereo file. Solo the sound, listen to see if there's any stereo spread in the low end. If not, it's fine to use as is. (Some kick samples will have a bit of stereo reverb, but the reverb generally has no low end to it, so it's not a problem) |
|
|
| G-Con |
| quote: | Originally posted by derail
G-Con, it's fine to leave it as a stereo sound if there's no stereo spread to the sound. A lot of stereo bassdrums and basses are actually mono, just in a stereo file. Solo the sound, listen to see if there's any stereo spread in the low end. If not, it's fine to use as is. (Some kick samples will have a bit of stereo reverb, but the reverb generally has no low end to it, so it's not a problem) |
Thanks for clearing that up :)
Reading the original posters post, I'm not sure if he thinks it is necessary to put the utility plugin on any track he wants mono when in fact ho doesnt need to do that at all |
|
|
| CReddick |
| quote: | Originally posted by palm
Render master at 24bit maybe 48kHz wav then let the label master it and encode it to mp3 16bit 44,1kHz
|
I have to disagree. You should work/render at a sample rate that is an integer of the final destination sample rate. I read an article from Roger Nichols that went over this in great detail. The mathematical algorithms that truncate the samples down can introduce noise when the math isn't even.
example, final 44.1khz
feel free to work at 44.1 (OBV), 88.2, or 176.4 |
|
|
|
|
