Is there any advantage to recording my dialogue in 24 bit, as opposed to 16bit? This would be for a horror movie.

+ safer level settings as you can leave more headroom
+ less rounding errors in complex mixdowns (mostly theoretical)

- takes up more space (not a big concern anymore)
- have to convert to 16 bits for editing anyway

Is there any advantage to recording my dialogue in 24 bit, as opposed to 16bit? This would be for a horror movie.

Noise in a signal is additive when mixing several tracks together. If you record in 16bit, by the time you get to the final output you may find that you have a S/N ratio equivalent to having recorded in 14 or 12 bit. By recording and mixing in 24 and going down to 16 when you master, you avoid that issue. That being said, other factors in your audio chain might make such considerations moot anyway.

Do any of the current NLEs support 24bit audio tracks?

Do any of the current NLEs support 24bit audio tracks?Oh yes, absolutely. Final Cut Pro, which is the one I'm familiar with, goes up to 96 kHz and 24 bits (more if you use Soundtrack Pro). I'd be surprised if this wasn't widely supported on Windows-based NLEs, too.

- Martin

Noise in a signal is additive when mixing several tracks together. If you record in 16bit, by the time you get to the final output you may find that you have a S/N ratio equivalent to having recorded in 14 or 12 bit. By recording and mixing in 24 and going down to 16 when you master, you avoid that issue. That being said, other factors in your audio chain might make such considerations moot anyway.


Well, noise is additive of course when you SUM the signals, but then also the usefull signal is additive. For this reason the level of summed signals has to be lowered to prevent overloading*. Lowering the general level brings down also the noise. Noise and "pay" signals are no different, system does not differentiate them (in analog systems the noise adds up each generation, but it is history). So I venture to say noise does not add up. '

Using 24 bits does give some advantages in effects processing etc. because there are rounding errors in manipulated signals which might take away the last one or two least signifigant bits. In reality very few of us can make clean 24 bit originals anyway (Deva or SD 700 series or Cantar type field recorders and fine mics), so this is mostly theory for us like Steve says.

*) example: two tracks mixed 1:1 (summed) with peaks at -2 DbFS and noise at -70 DbFS would result in +4 dBFS peak -64 dBFS noise file, not possible. Have to lower levels 6 dB to keep peaks at -2 dBFS, this also pushes noise back to -70 dBFS.

Oh yes, absolutely. Final Cut Pro, which is the one I'm familiar with, goes up to 96 kHz and 24 bits (more if you use Soundtrack Pro). I'd be surprised if this wasn't widely supported on Windows-based NLEs, too.

- Martin

Sony Vegas has supported 24/192 since the first version (nearly 8 years), even their cheapo Movie Studio supports it.
Adobe Soundbooth supports it.
I believe Canopus Edius supports it.

In other words, most NLE's support at least 24/96. But you're fine recording at 24/48, which I believe is supported by everyone.

Well, noise is additive of course when you SUM the signals, but then also the usefull signal is additive. For this reason the level of summed signals has to be lowered to prevent overloading*. ....

You are thinking analog mixing here. Digital mixing, such as mixing music, dialog, and FX tracks in an NLE, is a different situation. There the noise level adds while the signal level does not, leading to an effective decrease in dynamic range to below that normally associated with 16-bit resolution. For best results, record in 24 bit even if final release is to be 16 bit.

You are thinking analog mixing here. Digital mixing, such as mixing music, dialog, and FX tracks in an NLE, is a different situation. There the noise level adds while the signal level does not, leading to an effective decrease in dynamic range to below that normally associated with 16-bit resolution. For best results, record in 24 bit even if final release is to be 16 bit.

Why does that happen?

I routinely record live music for distribution on either DVD or CD. I've found that 16 bit recordings don't have the dynamic range necessary for live recordings( I get a lot of clipping of vocals), in close proximity and in an enclosed space. 24 bit recordings really allow me to record live, without as much input compression or limiting, so, it saves me quite a bit of trouble during setup, not to mention adding compression in post to fix clipped passages.

Why does that happen?

It has to do with the math involved in digital signal processing amd I'm not enough of a mathematician to properly answer that - I only know that's how it works. Perhaps one our resident signal processing engineers will chime in and correct my crude explantion.

I think it goes roughly something like this ... When you mix digitally you are not summing signal voltages, you are adding numbers that represent the quantized waveforms. First, in binary arithematic 0+0=0, 0+1=1, 1+1=10, 10+1=11, 11+1=100, 11+11=110, etc. Now take two signals just at clipping, the top of the dynamic range spectrum. They are each represented by '111111111111111' (16 ones). Adding them should result in '10000000000000000' (1 followed by 16 zeros) but because 16 bits is the maximum, the combined signal is also at '1111111111111111', still 16 ones. But smaller numbers, down at the noise level, add just fine without that overflow and truncation.

If you are recording in 16-bit and are releasing in 16-bit without mixing multiple tracks, no problem. But mixing multiple 16-bit tracks costs you resolution for each added track. Of course there's nothing magic about 16 bit, the same loss happens when recording and mixing at 24 bits. But there, because you're going to reduce the resolution to 16 bits for release anyway, the loss of dynamic range that drops that of 24-bit originals down to that equivalent to 22 or 20 or 18 bit recordings is obscured by the final conversion drops it farther down to that of 16 bit.

Using binary does really make no difference, math is math.

I see it this way: we have two strong signals and sum them (mix them together). If there is a danger of going over the limit (be it 16 or 24 bits) we have to lower the levels of both signals before summing, because after summing and causing clipping they can not be saved. By deviding both signals by 2 (shifting one bit to the right) we make sure the result can not be larger than one original signal alone. This actually means lowering the levels 6 dB. This also lowers the noise levels one bit to the right or 6 dB. If we call noise the low level part of the signal that should not be there (hiss from amps, mic etc) it is also first lowered in level and in summing returned back to what it was. I can not see how this math would affect only loud part of the signal and not the quiet parts in equal effect. Thus I do not understand why only the noise would add up and anything else not. The system does not know which part of the signal is noise, which is not.

If we call noise the dithering etc stuff which lives down there among the last bit realm (lowest few decibels) my expertize can not figure what happens there, if that part behaves in linear fashion or do quantum physics come into play.

16 bit signal can carry 96 dB dynamic range, it is certainly possible to fit all music, not to mention dialogue in that space (and nobody has reproduction systems where the dynamic range including the listening space backround noise exeeds this). If the loudest signals clip the levels are set wrong. This does not mean using 24 bits for recording is vanity, it does give some peace of mind as you can set the levels lower for safety. As long as the mic preamps and mics are good enough, that is, many times thay are not. If there is too much hiss down there to prevent using 16 bits for recording without constant fear of clipping, using 24 bits is not going to help with same mics, pre-amps etc. The noise floor is going to be exectly the same distance from the clipping point.

Now take two signals just at clipping, the top of the dynamic range spectrum. They are each represented by '111111111111111' (16 ones). Adding them should result in '10000000000000000' (1 followed by 16 zeros) but because 16 bits is the maximum, the combined signal is also at '1111111111111111', still 16 ones. But smaller numbers, down at the noise level, add just fine without that overflow and truncation.

Steve,

if you add '111111111111111' (16 ones in binary notation, or 65535 decimal) to itself (or multiply it with two, which is the same thing), the result is '11111111111111110' (16 ones and one zero, or 131070 decimal), which is way above the limit for 16 bits. The result you mentioned is what you'd get by adding just '1' to the original number, which is so darn close that - while technically clipping - it would sound alright.

If you sum up two tracks - digitally or analog - the result is naturally louder than the original tracks. If listen to two instruments, it is of course louder than just one of the two at a time. So if the tracks that you want to add up are close to the maximum level, you need to lower them before you add them, as Petri explained.

- Martin

Is there any advantage to recording my dialogue in 24 bit, as opposed to 16bit? This would be for a horror movie.

If you have the option to go 24 bit, do it. With 16 bit make sure you have a good limiter and go for a hotter signal (-12 or so).
There is an interesting comparision of 16 and 24 bit on the Sound Devices Website (www.sounddevices.com). Listen to it. Bottomline (robustness in post left aside): for reasonably hot signals you will not notice much of a difference, but for lower levels its a huge step up with 24 bit.

In a practical sense, it won't make any difference. I'm guessing that your movie does not have a Hollywood budget and is a do-it-yourself kind of thing. Your biggest problem will likely be making sure there isn't too much background noise. Most locations you shoot in will have some sort of background noise to contend with. Usually the loudest noise will hide the quieter forms of noise.

*Background noise- for narrative work, this can be very high. This is by far your biggest problem.
*Preamp noise (sometimes this is louder than the quantization noise in 16-bit equipment, so there would be no point in going 24-bit; but some equipment is really good, so there may be a reason to go 24-bit)
*Noise in the playback equipment
*Quantization noise. Shooting 24-bit lowers this to negligible amounts.

If you have any ambiance in your mix, it will effectively mask any quantization noise. So I wouldn't worry about it at all.

The biggest problem with sound in independent films are (in roughly the following order):
*Bad script. (Ok this doesn't count.)
*Too much background noise
*Too much reverb

*Poorly-done ADR (and some actors aren't very good at ADR).
*Bad mixing; Lav-recorded sound doesn't match other dialogue
*Music not that good (Again, doesn't really count.)

2- To deal with background noise, the best way to record it would be to:
--Record boom audio. Indoors, use a hypercardioid microphone (not a shotgun, especially not the cheaper ones). A decent boom operator helps.
--On seperate tracks, record wireless lavs. Getting the mic so close will dramatically reduce background noise. However, you will need to EQ the lavs + add reverb to make them sound right. The lavs also cost money and suck up batteries.
Recording multi-track is easier with two audio people (one doing boom, the other mixing).

3- In music going 24-bit can help a bit... but that's when your background noise can get real low (relative to the signal, which is louder than dialog).

Steve,

if you add '111111111111111' (16 ones in binary notation, or 65535 decimal) to itself (or multiply it with two, which is the same thing), the result is '11111111111111110' (16 ones and one zero, or 131070 decimal), which is way above the limit for 16 bits. The result you mentioned is what you'd get by adding just '1' to the original number, which is so darn close that - while technically clipping - it would sound alright.

If you sum up two tracks - digitally or analog - the result is naturally louder than the original tracks. If listen to two instruments, it is of course louder than just one of the two at a time. So if the tracks that you want to add up are close to the maximum level, you need to lower them before you add them, as Petri explained.

- Martin

Thanks - I knew the number didn't look right as I typed it but I was too sleepy to figure out the right one. Nonetheless, I'll still stick to my guns ... digital mixing can lead to a loss of effective dynamic range and recording at a higher bit depth can compensate for that. See also this discussion on digital recording in "Sound for Film and Television", Thomlinson Holman, page 64, where he writes "With any degree of mixing (of multiple 16 bit tracks) it is impossible to deliver an output that has a 16 bit dynamic range."

Just one small correction, the maximum value in a digital recording is represented by 0, not (2^n) - 1 where n is the bit depth. This makes for much simpler conversions between bit depths.
Certainly mix at 24 bit will be better than mixing at 16 bit however it's very common practice for modern mixinf apps to run their internal pipelines at more than 16 bits anyway so unless you're render back to 16 bit through multiple generations you don't gain anything.

Of course the dialog world got along quite nicely with 16-bit, 48 kHz DAT machines before becoming obsolete.

Regards,

Ty Ford

And before 16 bit DAT hissy analog tape was good enough...

Getting the most advanced gear and technology does not help if techique is not right.

And in the end many videos/films are delivered on DVD:s with MPEG sound not even 16 bit quality...

Finland, Australia, Canada, USA, etc. What a great forum!

16-bit for dialog is fine IF..notice the big if...the audio is particularly well recorded. However, for some members and visitors, audio is a necessary, mysterious PITA.

They can't afford the gear or a sound person and/or do not understand "best practices" to achieve well recorded audio themselves. For them, 24-bit may afford them some safety margin if there's someone who knows how to dig out the dialog with intelligent use of good EQ. In most cases, though, they aren't going to spend the time or money, so the point is moot.

I'm not trying to sound harsh here, but although I'm driven to get my clients the best possible audio (because that's what they pay me to do) many forum members and visitors aren't making a living with video and audio. For them, really good audio (or video) is not as important as it would be for me.

The MPEG audio used in HDV cameras is an interesting example. It has 1/5 the data of 16-bit 48 kHz audio. Some folk think that's fine!

Regards,

Ty Ford

...
16-bit for dialog is fine IF..notice the big if...the audio is particularly well recorded. ...
I'm not trying to sound harsh here, but although I'm driven to get my clients the best possible audio (because that's what they pay me to do) many forum members and visitors aren't making a living with video and audio. For them, really good audio (or video) is not as important as it would be for me.

The MPEG audio used in HDV cameras is an interesting example. It has 1/5 the data of 16-bit 48 kHz audio. Some folk think that's fine!

Regards,

Ty Ford

Agree with you 100%, Ty.

I have a lot of yelling and screaming in my movie, will 24 bits better for such loud signals?

I have a lot of yelling and screaming in my movie, will 24 bits better for such loud signals?

The dynamic range we're talking about is the ratio between the softest sound possible and the peak levels. Screaming and yelling are better controled in the analog realm with attention to level setting and proper application of limiting. Once the audio is pristine, there *might* be a subtle difference between 16 bit and 24 bit but going to 24 won't be a panacea for out of control levels.

Hey there, sorry to bring up an old thread but I'd like to add a few points to the 16 vs. 24 bits here. I've worked as a mastering, mix engineer and producer for the last eight years and just getting into video for the fun of it... so I'm a video newbie but an audio expert.

Something that a lot of guys don't realize, even in the audio industry, is what actually happens with your audio in the DAE. Always remember to apply the correct dither at the end of your audio chain. It doesn't matter if you have 24bit, 16bit or 8bit audio at the source. As soon as you start manipulating the audio in your DAE, using panning, volume, plugins, fades etc... you have 32bit float at your output bus. Just lowering a channel 0,1db means re-calcualting the audio, and to keep it as close to the original as possible the software needs 32bit float resolution or higher. If you have an audio project that you're mixing down to 16bit without adding dither noise at the output, it will sound horrible compared to the same project properly dithered. If you've ever had the experience of fading out a piece of audio, and that it didn't sound totally smooth... like at the end of the fade it sounds more like the sound just drops out rather than smoothly disappear, you've not dithered your audio correctly!

ALWAYS REMEMBER TO DITHER!!!!!

My take on dialogue recording is that it might of course sound better at 24bit compared to 16bit, but it also depends A LOT on the quality of your mic, preamp and A/D converters. 24bit also sounds a lot better if you need to apply heavy processing at some point, especially dynamic processing. There really is no excuse for NOT recording at 24bit if you can, just remember that unless you're using only this recording in your final production totally unprocessed... you will always have the need for DITHER!

Let's say we have a console with some three bit A/D converters with step size 1 volt (to make the math easy). If these converters are "offset binary" they will produce code 000 for -4 volts, 001 for -3; 010 for -2; 011 for -1; 100 for 1; 101 for 2 and 111 for +3 volts which is the largest positive number which can be represented with 3 bits and 1 volt per count. When these numbers are to be used for arithmetic operations they are converted to "2's complement" by changing the most significant (leftmost) bit so now -4 volts encodes as 100; -1 volt as 111; 0 volts as 000; +3 volts as 011 and so on. Note that these numbers are representations of the voltages at the input to the A/D which voltages are proportional to the air pressures at the microphone diaphragms. Thus the numbers are scaled approximations to sound pressure levels over time.

Now if we want to sum the output of 2 of these A/D converters we may be summing -1 with +3 to get +2 or +1 with +1 to get +2 or -2 with -2 to get -4 but there are equally good chances that we will be summing +2 with +3 to get +5 or -3 with -4 to get -7. Neither of these last 2 is representable with 3 bits so we must go to a 4th bit and indeed if we know we are summing three bit "signed integers" we will recode the numbers by duplicating the sign bit thus -4 becomes 1100; -3 = 1101, -1 = 1011; 1 = 0001 and so on. Now if we need to add -3 to -4 we write 1101 above 1100 and add the bits by the binary rules (1 + 0 =1; 1+1 = 0 carry 1) so 1101 + 1100 = 1001 (the carry into the 5th binary place gets discarded) which is -7 in 4 bit 2's complement notation (-6 is 1010 -1 is 1111; +7 is 0111...). So in adding 2 3 bit channels we need a 4th bit. If we were to add 2 of these 4 bit sums of 2 we would need a fifth; adding 2 sums of sums a fifth and so on. Thus every time you double the number of channels you need an extra bit but note that this is the most conservative approach. Signals and noises are usually uncorrelated meaning that it is the square of the voltage of the sum which doubles if two signals are added. Nevertheless the math requires the extra bit and it is therefore supplied. To require 24 bits as a consequence of summing 16 bit channels means that up to 256 channels could be accomodated.

If you sum 256 channels of 16 bits each forming a 24 bit sum and then go back to 16 bits by truncation or even rounding that is akin to quantizing the original signal to 16 bits and quantizing noise will be generated. It will be 10.8 dB below the least significant bit but if the signal is narrowband (unlikely from a sum of 256 sources) it will be correlated with the signal and you can hear it. Dithering decorrelates the quantizing noise from the signal but with a mix of 256 sources that shouldn't matter much. Dithering is also beneficial in this regard in the hardware A/D converters.

hi, i also wanted to chime in here. I use to ask the same question about 16/24bit. After a few productions and recording sound seperately i realized the huge difference. 24 bit gives you less noise and higher headroom, you also have more freedom when processing the files with filters and effects. When i mix my productions, there's so much room to play with. watch this weird short film we did and notice the yell at the end, its so clean and undistorted, there was also no gain riding at all. All SFX were 24bit too. Recored using the 416 mic and the Sound Devices 702T.

http://www.youtube.com/watch?v=jFWP1hGORis

The MPEG audio used in HDV cameras is an interesting example. It has 1/5 the data of 16-bit 48 kHz audio. Some folk think that's fine!

Ty Ford

And much of the time it is. If the entropy (a mathematical measure of the information content in a signal) is less than 16/5 bits per sample it is theoretically possible to convey all the information in the signal with 3 1/5 bits (average obviously) per sample. In other words compression in and of itself is not necessarily a bad thing. Most sources can be compressed by about 50% losslessly meaning that a 16 bit signal reconstructed from a 8 bit compressed samples is identical to the 16 bit original). The problems occur when "compression artifacts" arise because of unusual patterms in the sound (and implementations of lagorithms that don't realize their theoretical potentials). It's sort of like the extra bits required for summing situation. Most of the time you don't need the extra bit but when you do, you do.

Spoken like a mathematician. :)

Herb Squire's test at a past AES conference in NY audibly demonstrated the bad effects of of data compression on audio. I was there. I heard it. The eye is a lot more forgiving than the ear when it comes to data compression.

Further, depending on the use of the audio, subsequent compressions - like when you make a regular DVD and the audio is then compressed as part of the mpeg stream - more data is lost.

Were we able to hear both the original audio (recorded well) at 24-bit and then be able to compare that with HDV audio, that would be telling. I recall some folks double recording (to camera and to 702T) and letting us know the 24-bit sounded better. Of course, part of the problem could have been the camera audio circuits.

In a audio/video world where 24-bit audio is normal (and 32-bit is not unusual), the compromises made by the industry to make and sell HDV are questionable. But then VHS won over beta and there are still a lot of audio cassettes out there. If recorded well, those cassettes sound better than many of the XM broadcasts due to XM's data compression.

Let me decide when I want to chuck the bits, A. J.. :)

Regards,

Ty Ford

Ty,

I'm not a mathemetician but rather and engineer which in some quarters would be deemed even worse. That confession made let me assure you that I am the last guy to try to impose bit chucking on you because I am well aware of the problems that arise when the theoretical isn't well mapped into practice. It seems that in any compressed piece that I listen too (from I-tunes for example) there is always at least one nasty artifact which usually occurs on some bit that should be brilliant like a trumpet blast, symbol crash or high C even though the rest of the performance is fine (to my tin ear anyway). I, and some of my colleagues, have been wondering what the future of recorded music is to be now that CDs seem to be fading from the scene and SACD's looking as if it isn't going to make it.

Nevertheless, if I wanted to send you an (uncompressed) audio file I would certainly run it through Stuffit (or some similar compression routine) first in order to save time and bandwidth on the Net. When you received this file you would un-stuff it (expand it) and receive an identical, to the bit, copy of the original file. This is an example of lossless compression. The compressor doesn't care whether the data is video, audio, pressure data, text or computer source code. It analyses the 1's and 0's and codes the data in such a way that the number of 1's and 0's sent is about the same as the true information content of the file.

A.J says it beautifully, and probably correct!

Thing is, the ear is very sensitive to artifacts. I can't understand how people can listen to 320kb MP3-audio for instance, and I'm still far away from being an audio snob. But when you combine video and audio you're also dividing your attention. For the most part, audio will seem to sound better when you're also focusing on what you see (a well known fact amongst top-plugin developers) so you can get away with a lot more than if you were focusing on audio only. Even music videos seem to enhance the production quality of music by a great margin. You might see a video and be impressed by the audio production, but when listening to the audio only you might decide that it was really quite un-impressive.

You can get away with more audio compression than usual if you're combining it with video. I typically don't react to 224kb mp3 audio sounding bad when also focusing on video, but somewhere around there is the limit for me.

16bit vs. 24bit recording then becomes kind of silly to discuss since if you're not bothered by the former, you will most certainly not be bothered by the latter. Most people don't even react to a truncated 16bit audio stream, and given the right situation and production I would probably not react to it either since it only really has a musical impact when the sound becomes more dynamic. Most non-audio people would react to a truncated classical recording once shown what to listen for I'm sure, but most people including professionals, would not react to it if it were rock, metal or smashed RnB which is essentially very non-dynamic and distorted by nature so to speak.

To be on the safe side... always record at the highest resolution and always dither for your end-medium. You never need to think about it... just do it! :)

i use to think about it too much and it hurt my head. 16bit audio dedicated recorders generally will sound way better than any camera, thats because they are built for that purpose. Cameras are generally optimized for image making and sound takes a back seat, just like on set, but thats a different topic. If sound is important to you, capture it at the highest res you can get. I'm currently mixing a short, and im amazed with how much room you have to play with 24bit files. and the noise level is so low, i rarely have to apply any noise reduction filters.
I can also stack a few filters if needed and the margin of error is eliminated.
Dont listen to the crazy directors who cry in post about the audio and not giving us the chance to optimize it to begin with.

But like the pros here told me, a 16bit recorder in the right hand will sound better than with an amatuer with a 24bit recorder. Email me if you need a 24bit audio file to play with. good luck

But like the pros here told me, a 16bit recorder in the right hand will sound better than with an amatuer with a 24bit recorder.

True. However, a 24-bit recorder in the hands of an amateur who only needs to know one thing (leave lots of headroom) will sound better than a 16-bit recorder in the hands of an amateur who "tries" to manage the levels.

true, yes of course. Point is tools are good in the hands of the right person.

A.J says it beautifully, and probably correct!

For the most part, audio will seem to sound better when you're also focusing on what you see (a well known fact amongst top-plugin developers) so you can get away with a lot more than if you were focusing on audio only.

Thankfully, this works both ways.

That's why Im trying to improve my audio skills!

Sony Vegas has supported 24/192 since the first version (nearly 8 years), even their cheapo Movie Studio supports it.
Adobe Soundbooth supports it.
I believe Canopus Edius supports it.

In other words, most NLE's support at least 24/96. But you're fine recording at 24/48, which I believe is supported by everyone.

Sadly, Adobe Premiere Pro CS3 only supports a maximum of 16/96.

Avid supports 24-bit audio.

Sadly, Adobe Premiere Pro CS3 only supports a maximum of 16/96.
No way! Really? That is a huge oversight - it's mind-boggling, how can it be used for serious work? "Sweeten" outside Premiere? Is 24bit automatically truncated to 16?

No way! Really? That is a huge oversight - it's mind-boggling, how can it be used for serious work? "Sweeten" outside Premiere?


Yeah, it's a real pain.

Is 24bit automatically truncated to 16?

Not even; for Windows WAV files it just refuses to load them.

Daniel are you sure about that? I had no problems handing off my 24/48 broadcast wav files to a production company that used Premiere as their NLE tool. Apparently it doesn't work as smoothly as FCP, but they were able to break apart the poly files I gave them and swap out my tracks with the audio we ran into the camera.

Wayne

Daniel are you sure about that?

I was, but I just tried it and it works fine. The reason for my mistaken impression was probably some other problem with the WAV file (open file lock?) and not because it was 24-bit. Thank goodness I can edit in native 24-bit now. Sorry for the noise, everyone. :)

Another DVI success story!

Thanks for your persistence, Wayne.

Ty Ford