When do you think it will be standard to mix a film at 96K or possibly 192K and have it stay at that sample-rate?
Will Feature Films ever make the jump to 96K rather than stay at 48K?
Also, when you record dialogue or effects or foley, do you record straight to 48K or do you record at 192 or 96 and then downconvert? And how do you do the downconverting?
96 kHz and 32-bit floating point aren't for the final mix. They're for intermediate processing. In the final mix you can't hear a difference between 48 kHz and 96 kHz (unless you're Batman or the mix was downsampled poorly or the 96 kHz mix contains ultrasound that's distorting in your speaker). Likewise, 20-bit is adequate to represent all of the 120 dB dynamic range you can ever get out of a real system (due to physics).
BUT, when putting a waveform through a lot of processing in a DAW, you want lots of headroom in both frequency and amplitude, to prevent unnecessary computation noise, aliasing, phase shift, etc. Like photocopying a photocopy, the better the quality of each copy, the less degradation there will be in the final product.
Actually, I don't feel the need of 96kHz mixes. Something that sometime I really need (and it's already here with digital cinema) is uncompressed audio. More than one time I found the (not so bad, for the time when it was born) Dolby data compression (the ac3 like compression that the DMU do for the MODisk) to be really annoying and changing a lot the sound. But in the end it really depends from the kind of movie; and actually, the real big deal is the sound system of cinema theater: 8 times out of 10, the cinema has wrong level, or outdated ampli and speakers, or wrong sound correction. Just my 2cent
I'd say its pretty impossible in the near term and pretty unlikely in the long term.
First off, sample rate depth has serious consequences for the way that DAW and digitial processing work, and mixing is a processor intense activity. In order for a process like an eq or summing mixer or distortion to work correctly, it has to do its process at double or 4x the rate it would at 48k. Ever actually try mixing something complex at 192k? Give it a spin and let us know how your computer holds up.
Also, have you done a double-blind listening test between 48k and 96k? Are you sure you can hear a difference? I can certainly set one up for you. What if I use 48k source sample rate material recorded outdoors and then encode each into AAC? Because another issue is the production process: Most audio for films is not captured at super high sample rates. Its captured at 48k, so unless you're asking your location sound dept to run at 192k then you're just upsampling the dialogue and pfx anyway.
Actually, that first point is the biggest one. You're crippling your computing power and you are doing it for what will amount to zero sonic benefit. This is doubly true when you consider that location source audio is captured at 48k, and the final distribution master is encoded in a lossy format.
In the end quality is about content. Mixing decisions are about story, not sample rates. I'm not trying to be harsh, I'm just trying to steer you away from the placebo mentality that makes Monster Cables so much money. Trust your ears, but only when your eyes aren't involved.
Good question, i'd like to A-B some 96k vs 48k myself.
Although from a theoretical point of view, a sample rate of 96kHz means the signal can carry frequencies up to 48kHz (http://en.wikipedia.org/wiki/Nyquist_rate), and we can't really hear above 20kHz (most of us are lucky to hear up to 17kHz!). Whereas a sample rate of 48kHz can, theoretically, carry signal up to 24kHz, which is still beyond our range of hearing. Also, i don't think any widely used speakers are capable of reproducing any frequencies over 20kHz.
As to whether a higher sample rate affects the frequencies we can hear, i can't say. It may be possible, but i think the benefits to a typical audience would be negligible.
On the upside, FX recorded at high sample rates won't lose all their high frequencies when they're pitched down! (as far as i know...)
"I'm a total stickler when it comes to quality."
So.....
you must have noticed the X Curve in every screening you have ever been to then?
and if thats such a worry do you also notice the sync difference between the front row and the back row in a theatre?
Some Blu-Ray titles are offered at 24/96. So far, they mainly seem to be live concert recordings.
Given that major movie theaters are playing back basically mp3 quality audio, and that the film festival circuit is even worse (Sundance only supports Stereo LoRo or LtRt encoding for a phasey LCRS mix!), I have resigned myself to the fact that 48kHz is just the standard.
If i can I will always record at 192 kHz, and I can still hear 17 kHz despite being considerably older than 27. I have students who are considerably younger than 27 who struggle to hear 13 kHz.
The higher sampling rate does make a difference when it comes to reverberation, as it works in a similar manner to an LFO (Low Frequency Oscillator), so inaudible frequencies affect audible frequencies.
However, all of the cinemas I ever go to are lucky if they have the projector focused regularly, never mind having the audio properly calibrated.
We will move to 192 kHz eventually, but there is so much that has to catch up first.
The first thing that I would love is a loudspeaker with a flat frequency response even up to 17 kHz, even my favourite Genelecs colour the sound.
I hear all of you on this one, especially the OP. I can also tell a difference... ...Here's the deal: I grew up nearly deaf (and inching towards deafness). Sound didn't matter until a surgery gave me back full function in one ear. The other is bad, but with special equipment, can work up to 89% efficient, which, I'm told is average human quality for my age. I can only suggest that my experience of sound in such a manner has allowed me to articulate greater differences in how sound is experienced. To that I add the following:
I play on both near field speakers, as well as on a full surround system that was professionally installed and engineered with microphone field balance (connecting to mics and testing at different fields of the room). I can tell a difference. However, it's not an articulated auditory response (not necessarily pure "Hearing" of the sound). It's more in the "Feel" of the sound. I use a USB enabled receiver, with 48, 96 and 192khz sample rates in WAV. With 96k there's a touch more entry feeling than 48, and the anticipated sound feels fuller, more complete; also, the roll into the next notes (I work performances and do live audio recording for video post production archival) is more fluid. However, this is only slightly more noticeable in 192khz, or, I believe, my specs are maxed (on the receiver) at 96 and it interpolates the data down.
The old 20khz human response is based on sound room in-ear otiology, which limits the sound to what you can hear in your ears alone, but hearing is done throughout the body, and processed by multiple systems in the brain.
If you're unhappy with the mixdown, check your dithering and antialiasing modes in your DAW. Do some reading on their balances. You might find a preset that works well for you. They sort of "Fake" the feeling of the sound, much like the blur\sharpening effects of photoshop do on photos. Play around with each audio style you work with. IF you have a compression GUI that will accept pure WAV and give you effects to choose from (like apple's compressor), you can encode several files and check the balance. I would invest in a USB surround receiver and\or usb playback device, then put your original WAV or full format files on USB, along with your mixed set. Know your receiver's limits (check it out online or send a support request with your questions to the manufacturer). Play Original sound, then a mixed sound in headphones, and repeat with each mix, making notes about how much closer they get. DO the same with near field speakers. Do the same with your wide room surround. Once you get "Close Enough" for your target output, check that mix, and create a preset if possible. This will give you the best results to your preference. Audition gives you only one output at a time. AME doesn't really allow you to add effects, and doesn't allow you to tweak the downsample. You're best off with a program similar to AME, or More like Compressor, that allows you to add basic audio effects to properly mix down or compress the range, with little perceptual loss.
Most audio systems are targeted to the ear, because it is the primary sensory organ for audio response. But most also include a SubWoofer that processes frequencies and builds a "Feel" of percussion or Force into the mix based on presets in the receiver. The better your subwoofer, the more natural and forceful the feel; but don't discount the receiver's algorithm, as it sends a processed range to the subwoofer. If you play a 96khz naturally recorded sound and a 48khz naturally recorded sound (both being the same sound), you'll not notice the difference on any really explicable level. IF you are so tuned, however, you might (6-28% average of professionals) be able to vocalize that the sound just "feels" different, and you'd be right. In recording digitally, you don't record all frequencies, normally. With this in mind as a hypothesis, I ran tests.
My conclusions: With 48khz samples, some of the lower band is dropped to favor the higher bands, stretching to include them and provide headroom for editing. With 96khz, more of the lower is processed but it's still stretched up. And with 192, the lower band is almost completely covered, and there is no shift, so that it doesn't stretch to higher bands. Beyond this, I haven't seen any of them attempt to stretch. The bottom is usually about 100hz-300hz for 48k and below. With 96 it's more like 50hz-200hz.
I tested this by producing a varying tone and checking the waveform that was recorded. Everything that didn't reach a certain decibel level (about -15db) was ignored at 300hz with 48khz, and lower than that, it was down to -9db. This is where the drop comes in. I was able to pick up sound at -40db at 350hz. Upon trying to adjust where there was apparently drop, I tried to see if there was anything there to view at all, and there was nothing but signal noise. I checked my hardware by inputing the sound at full output volume into the capture device, which showed input even at 0 attenuation, then used it to input by usb into the computer across 48 and 96khz files. Same behavior. I had to attenuate to catch the sound and have the attenuation almost a quarter up to pick up anything (though not even audible). I then played the files back on my surround system, and noticed there was some vibration of the drivers on the subwoofer at a basic level (about 15 on it's volume). This is a "normal" level for basic ear hearing (according to the documentation it's the volume where you should be able to hear the sound in that size room if you have normal hearing), and didn't really "Feel" like much. I raised the volume slowly and got signal noise, but as I raised the volume I began to hear a smoother tone (up around the max, which was 40-50 on volume attenuation).
This last test proves that those frequencies are played, even if we don't exactly "Hear" them. With a 96k playback I could actually feel a bit of a smooth effect in the sound, under the sound really. Does it prove everything? No. It does hint at the engineering of the hardware and software. Maybe my DAW is trying to allow for the sharp and shrill being balanced (vocals and high instruments that have a more "cutting" feel), and it's only filling the base or lows that can be played by modern hardware.
By the way, for future reference, the sample rate with digital audio has more to do with cutting up the sound rather than the frequency; however, since sound is a wave phenomena that can be graphically represented by plot points with numeric value, cutting it up does offer a way to capture the frequencies at a given instant. The difference is that everything in between is Guessed, as there is not solid algorithm for how the sound molds in between the two plotted points. The rule of thumb is: More plotted points, more accurate graph of the sound. With a continuous tracking (not sampled but continuously captured), the sound would be a nearly perfect representation, if such a tracking method could be done. However, tracking and plotting the points would be an infinitesimal task, occurring an infinite number of times per second. Since this isn't possible, we attempt to reproduce the sound using the wave phenomena as a guide, sampling the wave at points along the propagation, with similar frequency to the wave itself. This has the following effect: we can reproduce a 48khz width of sound with some accuracy, but as the sounds are piled together across frequencies, the samples don't perfectly match up and some of the information is lost, making the transitions sharper and less curved\smooth (graphically). When you increase your kHz, you can reproduce the sound more accurately, and capture the higher frequencies, but only if, at some point, you process and adjust the capture to pick up the right point of that frequency (the high and\or low of the wave) rather than an in between. Thus, you'll almost never be perfectly accurate. You'll lose quality when you drop your kHz, mainly because you're dropping the in-between points off the graph, and you'll find that your sound really loses more "Feel". However, by dithering and adjusting the Aliasing of the drop, you can provide a curvature to the interpolation, and "Fake" the feel a little. The samples on the recorder have nothing to do with the kHz of microphone or the sounds actual frequency, other than that you can sample at the same speed as the waves, and mark the amplitudes you measure. However, nothing says those amplitudes are right on the mark for the peak and trough of the actual wave. There is already a drop in quality. Add the fact that those measurements are plotted, and then connected crudely, and you can understand that its not the frequency you are measuring, its the amplitude of each interpolated sound frequency at that particular sampling moment (and they are all split into their respective ranges with varying accuracy of numerics; your bit width). For example, I can capture a 96khz sound at 48khz, and if it is a continuous tone over two or more seconds, it will sound almost the same when interpolated. If it varies at all, it will lose that variation quality. When playing back analog audio, the amplitudes are represented for the different frequencies over continuous waves. With digital, there is a conversion to a continuous format for the drivers to represent, as the frequencies are all split up and have to be mixed, and there is some shifting or inaccuracy of the timing. This is why most say that a Vinyl LP still sounds better than a CD.
In my subjective tests, I can't hear much/any difference between 96 and 48k. 24 and 16 bit however is a different story.
If I were you I'd be more worried about audio compression than sample rate.
No way I could tell the difference between 48 and 96 myself, this might be my old battered ears but I doubt it. However I can tell if someone's used a wank compression algorithm.
