24/96 Audio files not being rendered by Vegas 13

cft wrote on 6/25/2014, 7:57 PM
I am trying to figure out why Sony Vegas 13 (Build 310, 64 bit) won't render 24/96 audio on the timeline. I put numerous 24/96 audio .flac files on the timeline and "render as" to Microsoft .wav with the following output parameters:

= 96,000 Hz, 24 Bit, Stereo, PCM

The numerous .flac files add up to 1.85 GB when added together. When I attempt to render this, Vegas gives me the following popup window:

"An error occurred while creating the media file "XXXXXX". The file being rendered has exceeded the maximum size allowed for the selected format."

Apparently, Vegas doesn't like large 24/96 files. Anyone know a work-around for this, or is this a bug?

Anyone know exactly what the largest 24/96 file that Vegas can render is?

I have a brand new, fast computer - my specs are in my profile. I don't think this is a hardware or computer limitation.

Thoughts, thanks!

Comments

john_dennis wrote on 6/25/2014, 8:05 PM
" I don't think this is a hardware or computer limitation."

It's likely a limitation on the size the output file can be. Select just a few minutes of the 24/96 audio and try to render that.

From wikipedia:

"The WAV format is limited to files that are less than 4 GB, because of its use of a 32-bit unsigned integer to record the file size header (some programs limit the file size to 2 GB).[26] "

1.85 GB of compressed FLAC would likely become more than 2GB of uncompressed PCM.
vtxrocketeer wrote on 6/25/2014, 8:23 PM
I run into this limitation all the time because most of my work is stage show recordings lasting 60-90 minutes, where I routinely render uncompressed 96/24 wav files. I just render to wav64, which is one of your audio choices in the Render dialog. I don't know (and I haven't cared) what program other than Vegas and DVD Architect will recognize and play wav64 files.
cft wrote on 6/26/2014, 2:47 AM
Ah, I didn't know about the limitations of Microsoft .wav. Thanks for the info gents! Looks like I'll be using wav64 from now on.
DeadRadioStar wrote on 6/26/2014, 6:14 AM
I'm just curious as to why you would want to do this .... FLAC is lossless and has less limitations, as well as smaller files. As you can see, you can also drop them directly on the timeline. It may be that you're just converting for export to another system, but bear in mind that many digital audio workstations can in the meantime now also directly work with FLAC. In any case, yes, Wave64 is the solution!
farss wrote on 6/26/2014, 6:21 AM
Plus why 96KHz, 24bit yes but 96KHz serves no purpose other than to use up disk space.

Bob.
vtxrocketeer wrote on 6/26/2014, 7:20 AM
Plus why 96KHz, 24bit yes but 96KHz serves no purpose other than to use up disk space.

Maybe the OP will respond to your question, but I routinely record in and use 96/24 for my Blu-ray versions of choral and orchestral projects, where sonic quality is the top priority. (Still, perhaps 5-10% of my listeners will discern a difference between 96 and 48kHz.)
farss wrote on 6/26/2014, 7:41 AM
[I]" I routinely record in and use 96/24 for my Blu-ray versions of choral and orchestral projects, where sonic quality is the top priority"[/]

There's no improvement in sonic quality between 48KHz and 96KHz. At 48KHz the Nyquist limit is at 24KHz which is well above human hearing. There's a risk with 96KHz of reduced audio fidelity if the recording is down sampled to 44.1 or 48KHz. Of course internally most audio recorders sample at 96KHz or above then they down sample using very good hardware, it's difficult to emulate the algorithms the hardware uses in software.

I think this was discussed recently here, it was explained many years ago on the audio forum and I'm sure there's a number of articles about this on the web.

Bob.
vtxrocketeer wrote on 6/26/2014, 9:36 AM
I'm humble enough to accept correction in public, and after reading some background I realize that I've likely proceeded on the basis of misguided though likely not harmful assumptions in recording at 96 or even 192 kHz: http://people.xiph.org/~xiphmont/demo/neil-young.html

In other words, Bob, you're correct. ;)
cft wrote on 6/26/2014, 2:36 PM
In regards to why I would use FLAC and why 24/96, it's very simple. I recently downloaded some albums in 24/96 Flac files and want to put them on Blu-ray so I can listen to the files on my surround sound system. Since my audio receiver can play them, I wanted to see if there was any difference as opposed to the CD versions which I also have.

John_Cline wrote on 6/26/2014, 6:34 PM
24 bit audio can certainly make a noticeable difference in quality with programming that has a wide dynamic range. I come from an audio background and work with a lot of "golden ear" types, particularly on jazz and orchestral projects, I consider myself to have a pretty discerning ear as well. I have never met anyone that could reliably tell the difference between 48k and 96k or 192k sample rates. That said, I have a lot of old direct-to-2track analog recordings made on my Technics RS-1520 reel-to-reel recorder, it has a frequency response that is down only -3db at 50Khz. In these recordings there is definitely significant musical energy in the octave between 20Khz and 40Khz and I do use a sample rate of 96k-24bit when transferring the recordings just to capture all of the available signal. Practically speaking though, it really makes no difference. A technical argument could be made that a higher sample rate produces two or four times more samples at the highest audio frequencies, but this is also of little practical use. The point is that in the real world, 24bit sample depth is far more important than sample rate.
PeterDuke wrote on 6/26/2014, 8:55 PM
To down sample from say 96 kHz to 48 kHz all you have to do is first low pass filter the source signal to below 24 kHz and then discard every other sample.

The steeper the roll-off you want (to maintain high frequencies below the Nyquist frequency, which is 24 kHz in this case) the more complex the filter has to be.

If you want to eliminate phase distortion you can use a symmetrical Finite Impulse Response filter, which will have a long response time and introduce a delay equal to half the response time.

If you are filtering in non-real time, this delay is of no consequence provided that you allow for it in synchronising with video or other signals.

You can also use an Infinite Response Filter which will be faster to process for the same cut off rate, but will introduce phase distortion. This can be cancelled out by reversing the audio and filtering again, with twice the cut-off attenuation of the single pass process.